Re: [asterisk-users] IAX Java Softphone?
On Wed, 2009-01-14 at 17:38 +, Tim Panton wrote: > On 14 Jan 2009, at 16:47, Matthew Rubenstein wrote: > > > Thank you for getting that code contributed to the community. Is > > there > > a spec somewhere of the features supported by that applet? A version > > history? Docs of the "SDK" it's distributed as? > > All I have is the link. > > I should emphasise that I no longer have any relationship > with Mexuar so I'm in the dark as to exactly what their plans are > as far as supporting this code is concerned. > I'm just one of the original authors and an open-source proponent. > > I guess it would make sense for someone to open a sourceforge project > for it > and add those things. Do you know if there are at least hooks in there for the applet to do video over IAX? > Tim. -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX Java Softphone?
Thank you for getting that code contributed to the community. Is there a spec somewhere of the features supported by that applet? A version history? Docs of the "SDK" it's distributed as? On Wed, 2009-01-14 at 14:38 +, Tim Panton wrote: > I'm delighted to be able to say that as part of the agreement on my > departure from Mexuar, > the Corraleta applet source code Westhawk Ltd wrote for them has been > released under the GPL. > > it is available for download at : > > http://www.mexuar.com/files/corraleta_sdk.rar > > > Tim. > > On 20 Sep 2007, at 18:48, Matthew Rubenstein wrote: > > >Does anyone know of an IAX softphone in Java, whether applet or > > application? Even the most minimum featureset, just voice and dialing, > > or even embedded in some other app/let. Preferably GPL. Thanks. > > -- > > > > (C) Matthew Rubenstein -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP vs. SKINNY
On Thu, 2008-06-26 at 06:15 -0500, [EMAIL PROTECTED] wrote: > Date: Wed, 25 Jun 2008 23:41:18 +0200 > From: Michiel van Baak <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] SIP vs. SKINNY > To: asterisk-users@lists.digium.com > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=us-ascii > > On 14:16, Wed 25 Jun 08, Joe Carroll wrote: > > Can anyone comment on the performance benefits when comparing sip to > skinny ? > > Most cisco phones work better with the skinny firmware. > > That is not true when connecting to asterisk though. > > It all depends on the version of asterisk you are running. > I have a setup with over 20 skinny phones on asterisk -trunk and that > works great. Specially after today, now that chan_skinny supports > transfers. > > If you are running 1.4 I'm not sure what is best. It basically depends > on what you are doing with the phones. > In my home setup it worked great, but in my business I have to run > trunk > for the phones to be as workable as the sip variant. > > The skinny firmware has some neat stuff like XML push etc. > Dont know how the current SIP firmware is doing, as I have not run it > in > over 2 years now. > > YMMV Does Skinny let Cisco 79xx phones act as extensions *across the Internet* to a remote Asterisk server? Does SIP? How do the different SCCP channels compare to the chan_skinny support, in Asterisk 1.6? Is there a better guide than http://www.voip-info.org/wiki/view/chan_skinny to getting chan_skinny working best with Asterisk and Cisco 79xx phones? > Michiel van Baak -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adding ;password=foo;method=bar to SIP uri
Is there any reason that the SIP INVITE URL shouldn't conform to the same syntax as RFC3986 standard URLs ( http://en.wikipedia.org/wiki/URI_scheme#Generic_syntax ), as specific to SIP according to RFCs 3969 and 3261? That would be, according to sip:[:]@[:][;][?] examples: sip:[EMAIL PROTECTED]&priority=urgent sip:+1-212-555-1212:[EMAIL PROTECTED];user=phone Like sip:xyz:[EMAIL PROTECTED];Authorization=bar+realm%3Dbaz OR sip:xyz:[EMAIL PROTECTED];?Authorization:+bar;realm%3Dbaz or something along those lines, as per http://tools.ietf.org/html/rfc3261#page-194 ? On Thu, 2008-06-19 at 03:38 -0500, [EMAIL PROTECTED] wrote: > Date: Wed, 18 Jun 2008 18:34:15 -0400 > From: "Tom Browning" <[EMAIL PROTECTED]> > Subject: [asterisk-users] Adding ;password=foo;method=bar to SIP uri > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > To send calls into a custom SER implementation, I need to be able to > add > some items to the URI that Asterisk will then send as part of the > INVITE > > > Asterisk dial SIP/[EMAIL PROTECTED] > > needs to become > > Asterisk dial SIP/[EMAIL PROTECTED];password=foo;method=bar > > This is not a registration password. It is a passsword associated > with the > destination xyz at location abc.com > > Asterisk 1.4.18.1 seems to glue the data as part of the hostname and > fail to > lookup abc.com > > Is there a way to manipulate the URI that will be sent in the INVITE > to > accomplish this? > > Thanks in advance, > > Tom -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Manager Telnet Times Out?
I use the Asterisk Manager API by telneting into localhost:5038 then issuing action directives (usually inside a script). The telnet connection connects in a few milliseconds, and usually the Action: Login (and other Actions, including arbitrary valid Asterisk commands like "show channels") take well under a second to complete. But sometime in the last few days the Message doesn't return for closer to 20-30 seconds, but it does return properly. I haven't changed anything else on the machine since it was working, practically nothing except Asterisk and its dependencies are running (not even any calls), CPU load is about user: 0.001%; kernel: 0.02%; io: 0.004%; idle: 99.97% . What could make the Manager take so long to turn around actions all of a sudden? I'm using Asterisk 1.2 on Debian 4.0 on a P4/3.2GHz/2GB on which about 1.95GB is free and there's no swapping or any other evident thrashing. I have restarted the machine, and the Manager performance isn't changing. -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No-mobo PC for USB Drives Enclosure?
On Wed, 2008-05-14 at 14:06 +1000, Col Ferguson wrote: > If I understand right, your problem is that the power supply won't turn on ? > ATX power supplies can be told to turn on by jumpering 2 pins on the > motherboard power connector. From memory its the Green wire and one of the > black wires, I usually use the next one inwards. Pinouts for the connector > can be found via Google. > If the power supply also has an external on/off switch you can jumper these > pins and use the switch to turn the power on or off. > > Hope this helps, Thanks, that sounds like exactly what I was looking for. Is there any safety risk from jumpering that sensor? Like some kind of extra sensor, like voltage feedback, temperature or somesuch. If this works, it might point to a good way to reduce redundant Asterisk servers, which suck power, by just plugging the drive from each old server into the USB of a single server with a merged dialplan and a few other tweaks to point at several different mounted drives, rather than one per host/IP#. > Col > > > > - Original Message - > From: "Matthew Rubenstein" <[EMAIL PROTECTED]> > To: "Asterisk -Users" > Sent: Wednesday, May 14, 2008 12:22 PM > Subject: [asterisk-users] No-mobo PC for USB Drives Enclosure? > > > > I have over a half-dozen different SATA hard drives, each with > > different data (configs, voiceprompts, voicemail, CDRs, AGIs) for each > > one's different user groups and applications. Each one's load on the > > Asterisk server is small enough that one server can host them all, > > accessed easily over USB. > > > > But right now, each one is in its own external USB enclosure on a > > powered USB hub. I want to combine them all into a single large > > enclosure. I tried to use a single PC chassis, leaving the USB hub > > inside with the drives screwed into it, and powered from the PC power > > supply as internal drives on the proper drive power output plugs. But > > without a PC motherboard plugged into the power supply, too, the power > > supply won't start up to power the drives. > > > > I don't want to add a motherboard: that costs money, and sucks power, > > and is totally unnecessary. I just want to make this gutted PC chassis > > power my drives only, and have them connect to the complete PC sitting > > next to it via the single USB cable coming out of the drive chassis. How > > do I do that? > > > > Is it possible to use the extra, unused floppy power plugs to power > > more hard drives, with an adapter? Is it possible to split the existing > > hard drive power plugs to each power multiple drives? How many drives > > can I split each power plug into? The power supply is a cheap 300W unit, > > and the drives draw max under 9W each: > > http://www.wdc.com/en/products/products.asp?driveid=311 . So can I power > > 25-30 of these drives, or at least 10? > > -- > > > > (C) Matthew Rubenstein > > > > > > ___ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > -- > > No virus found in this incoming message. > > Checked by AVG. > > Version: 7.5.524 / Virus Database: 269.23.16/1430 - Release Date: > 5/13/2008 7:31 AM > > > > > -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No-mobo PC for USB Drives Enclosure?
On Tue, 2008-05-13 at 22:46 -0400, Steve Totaro wrote: > On Tue, May 13, 2008 at 10:22 PM, Matthew Rubenstein <[EMAIL PROTECTED]> > wrote: > > I have over a half-dozen different SATA hard drives, each with > > different data (configs, voiceprompts, voicemail, CDRs, AGIs) for each > > one's different user groups and applications. Each one's load on the > > Asterisk server is small enough that one server can host them all, > > accessed easily over USB. > > > > But right now, each one is in its own external USB enclosure on a > > powered USB hub. I want to combine them all into a single large > > enclosure. I tried to use a single PC chassis, leaving the USB hub > > inside with the drives screwed into it, and powered from the PC power > > supply as internal drives on the proper drive power output plugs. But > > without a PC motherboard plugged into the power supply, too, the power > > supply won't start up to power the drives. > > > > I don't want to add a motherboard: that costs money, and sucks > > power, > > and is totally unnecessary. I just want to make this gutted PC chassis > > power my drives only, and have them connect to the complete PC sitting > > next to it via the single USB cable coming out of the drive chassis. How > > do I do that? > > > > Is it possible to use the extra, unused floppy power plugs to power > > more hard drives, with an adapter? Is it possible to split the existing > > hard drive power plugs to each power multiple drives? How many drives > > can I split each power plug into? The power supply is a cheap 300W unit, > > and the drives draw max under 9W each: > > http://www.wdc.com/en/products/products.asp?driveid=311 . So can I power > > 25-30 of these drives, or at least 10? > > -- > > > > (C) Matthew Rubenstein > > > > Is the reason for separate drives security or something else? How > much data will the max size drive hold? > > Maybe a few of these could solve your problem? > http://www.buy.com/retail/product.asp?sku=206821004&adid=17070&dcaid=17070 > > Looking for a JBOD SATA enclosure with six slots but they are way expensive. The drives are 750GB drives, each one a different related set of apps from a different Asterisk machine. I've consolidated them all into a single Asterisk server. And I already have the existing PC chassis and power supply, as well as the $10 each SATA/USB adapters. If I can just figure out how to power them from the PC power supply without plugging in a useless motherboard, I'll have it done without spending any money (other than whatever cheap part tells the power supply to run without a mobo). > Thanks, > Steve Totaro -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No-mobo PC for USB Drives Enclosure?
I have over a half-dozen different SATA hard drives, each with different data (configs, voiceprompts, voicemail, CDRs, AGIs) for each one's different user groups and applications. Each one's load on the Asterisk server is small enough that one server can host them all, accessed easily over USB. But right now, each one is in its own external USB enclosure on a powered USB hub. I want to combine them all into a single large enclosure. I tried to use a single PC chassis, leaving the USB hub inside with the drives screwed into it, and powered from the PC power supply as internal drives on the proper drive power output plugs. But without a PC motherboard plugged into the power supply, too, the power supply won't start up to power the drives. I don't want to add a motherboard: that costs money, and sucks power, and is totally unnecessary. I just want to make this gutted PC chassis power my drives only, and have them connect to the complete PC sitting next to it via the single USB cable coming out of the drive chassis. How do I do that? Is it possible to use the extra, unused floppy power plugs to power more hard drives, with an adapter? Is it possible to split the existing hard drive power plugs to each power multiple drives? How many drives can I split each power plug into? The power supply is a cheap 300W unit, and the drives draw max under 9W each: http://www.wdc.com/en/products/products.asp?driveid=311 . So can I power 25-30 of these drives, or at least 10? -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk ZRTP?
What's the status of ZRTP supported by Asterisk? There was some discussion on the -dev list and -users list, but it was inconclusive. At about the same timeframe, a bug (#0010024) was opened and updated for several months, but has been "suspended" since late 2007. Does any version (1.4.x, 1.6.x) of Asterisk support ZRTP with clients (or with other servers)? Any successful testing with specific clients/peers to report? If not, are there any serious efforts underway? http://www.google.com/search?q=site%3Ahttp%3A%2F%2Flists.digium.com% 2Fpipermail%2Fasterisk-dev%2F+zrtp http://www.google.com/search?q=site%3Ahttp%3A%2F%2Flists.digium.com% 2Fpipermail%2Fasterisk-users%2F+zrtp http://bugs.digium.com/view.php?id=10024 -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ATA for Fax with BroadVoice?
I've got Asterisk 1.4 running on my LAN, with a BroadVoice account over my cablemodem. I've got an HP fax/printer/scanner. What's the cheapest ATA I can use to most reliably send and receive faxes from the HP as a fax machine? Should I config the ATA to fax directly to BroadVoice, or will I have more reliability sending it through some app on my local Asterisk, and then from Asterisk to BroadVoice (and then to the PSTN) - and maybe for receiving faxes, too? Or maybe I can use the HP as a printer/scanner over USB to the Asterisk box, without an ATA, and use some Asterisk SW (or other Linux app) to send/receive the fax images via BroadVoice. Or maybe there's some other Internet fax gateway I can use either my fax/ATA or Asterisk, so I can use my cablemodem for this fax send/receive work. -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Tests in VMWare (was: Re: asterisk-users Digest, Vol 44, Issue 104)
On Mon, 2008-03-31 at 03:04 -0500, [EMAIL PROTECTED] wrote: > Date: Mon, 31 Mar 2008 07:55:08 +0300 > From: Tzafrir Cohen <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Tests in VMWare > To: asterisk-users@lists.digium.com > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=us-ascii > > On Sun, Mar 30, 2008 at 08:50:10PM -0400, Ein Bielaczyc wrote: > > I'm just wondering if any one else has tried to successfully install > > Asterisk on Ubuntu inside VM. > > What version of Ubuntu? What version of Asterisk? They're not allowed to tell you: > NOTICE: This E-mail (including attachments) is covered by the > Electronic Communications Privacy Act, 18 U.S.C.2510-2521, is > confidential and may be legally privileged. If you are not the > intended recipient, you are hereby notified that any retention, > dissemination, distribution or copying of this communication is > strictly prohibited. Please reply to the sender that you have received > the message in error, then delete it. Hell, I wasn't even allowed to tell you that they're not allowed to tell you. > -- >Tzafrir Cohen -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Telemarketer Torture.... (was: Re: asterisk-users Digest, Vol 44, Issue 49)
You could accept as the "passcode" the caller punching in their own phone#, then checking that against your whitelist. Lets associates get past the challenge when using someone else's phone, without their remembering some arbitrary passcode. And strangers or barred old associates who abuse it can get an earful about how you're suing them for wire fraud. Preferably after you transfer them to an extension that plays a recording asking for their current calling#, so "you" can call them right back, and then the script threatens them. Automatically emailing your district attorney with their contact info optional. On Sun, 2008-03-16 at 12:00 -0500, [EMAIL PROTECTED] wrote: > Date: Sun, 16 Mar 2008 14:37:00 + > From: Horwich IT Services (Godwin Stewart) <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Telemarketer Torture > To: asterisk-users@lists.digium.com > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=US-ASCII > > On Sun, 16 Mar 2008 08:50:50 -0500, Lyle Giese <[EMAIL PROTECTED]> > wrote: > > > I just forward them to one of those two extensions. If callerid > worked > > more reliably I would automate it. But I get a lot of caller id > failures > > on my incoming POTS lines, esp when calling in from my cell phone. > > The way I worked around this problem was to give a passcode to people > I want > to hear from even if they conceal CLI. > > If an inbound call comes in without CLI (or with CLI but the number is > in > my blocklist for that matter), I forward it to a recorded message > saying > "Caller ID screening is in operation. Please press 1 if you are an > authorized caller". When the user complies, they're prompted for the > passcode. If it's correct, then the call is forwarded to my extension. > > Those I do want to hear from are not just blown off, they have a > chance to > get through to me regardless of the screening. Teleslime doesn't, and > they've paid for the call anyway. > > -- > Godwin Stewart - Horwich IT services -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LDAP (was: Re: asterisk-users Digest, Vol 44, Issue 48)
If you write a HowTo, would you please insert it into the wiki at http://www.voip-info.org/wiki/index.php?page=LDAP ? Thanks. On Sun, 2008-03-16 at 07:09 -0500, [EMAIL PROTECTED] wrote: > Date: Sat, 15 Mar 2008 18:20:32 -0200 > From: "Gonzalo Servat" <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] LDAP > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > On Fri, Mar 7, 2008 at 9:52 AM, Faraz Khan <[EMAIL PROTECTED]> > wrote: > > > It does work. Did you do the switch statement in extensions.conf? > > > > If not check voip-info for "Asterisk Realtime Extensions" > > > > Hi Faraz, > > I just realised I never replied to this message. Yes, you were right. > I > simply had to add "switch" to the right context and it worked > smoothly. > > I've actually managed to get it setup the way I want it (I'm going to > write > a HOWTO when I get a few minutes on how I did it, for the next > person). I > just managed to get VoiceMailMain() and Voicemail() to work straight > from > LDAP which is way-cool. I was wondering if you know (or if it's even > possible) to set the different voicemail settings that one can > normally set > in voicemail.conf into LDAP (I'm talking about things like the user's > voicemail password, email address for sending voicemails and the last > column > that specifies the different voicemail switches). > > Thanks very much again for your help!! > > Best regards > Gonzalo -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gartner Article (was: Re: asterisk-users Digest, Vol 44, Issue 32)
'Put even more simply by [Gartner analyst] Dulaney: "I'll do anything for money."' That, in a nutshell, is why Gartner is so out of touch. It pays! On Tue, 2008-03-11 at 12:00 -0500, [EMAIL PROTECTED] wrote: > Date: Tue, 11 Mar 2008 12:12:06 -0400 > From: "Dean Collins" <[EMAIL PROTECTED]> > Subject: [asterisk-users] Gartner Article > To: > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > Shows you how out of touch Gartner can be with reality at times; > > http://searchcio-midmarket.techtarget.com/news/article/0,289142,sid183_g > ci1304670,00.html?track=NL-973&ad=629348&asrc=EM_NLN_3233081&uid=1562002 > # > <http://searchcio-midmarket.techtarget.com/news/article/0,289142,sid183_ > gci1304670,00.html?track=NL-973&ad=629348&asrc=EM_NLN_3233081&uid=156200 > 2> > > > > "publishing reports weighed by the kilo as my old boss used to say" > > > > > > Regards, > > Dean Collins -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] replace astdb with a cluster-capable sql database engine (was: Re: asterisk-users Digest, Vol 44, Issue 22)
unix-odbc with Asterisk Realtime is one good way to use a different backend DB than MySQL. I haven't heard of "bit rot" problems running it over long times, but I'd like to if there are any. I'm particularly interested in seeing reports of Asterisk Realtime backed by Postgres. The problem with pointing dialplan DB functions like Set(DB) at unix-odbc (or any relational driver) is that the native functions use the very fast BDB, not a relational one, that has very different (better) scaling profiles than running those calls over a database driver, especially across a network. Having all those BDB data available in the relational DB for joins and other integrated queries (and backup and other RDBMS features) would be great, but there is danger in switching from the simple and high performance BDB into a more complex RDBMS. One way to do it is to leave the native BDB system, but interface a replica in the RDBMS to it. A polling process that replicates the BDB data into the RDBMS, and (if not negligible) updates the RDBMS with a read whenever the RDBMS copy is used (and then writes to the BDB when the RDBMS replica changes) would let the BDB remain as a fast/reliable "cache" directly to Asterisk, but use its data properly in the RDBMS. I'm interested in seeing any work performed on integrating Asterisk's data tier away from its defaults. Especially when that work is making Postgres the authoritive data store. I have various info that can help such a project, if people are really working on it. On Sat, 2008-03-08 at 20:08 -0600, [EMAIL PROTECTED] wrote: > Date: Sat, 8 Mar 2008 10:01:28 -0800 (PST) > From: Vieri <[EMAIL PROTECTED]> > Subject: [asterisk-users] replace astdb with a cluster-capable sql > databaseengine > To: asterisk-users@lists.digium.com > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=iso-8859-1 > > I've been searching the Internet for information > regarding the replacement of astdb with a modern sql > engine. > > There are several reasons one would like to do this. > First of all, external applications have a hard time > reading/writing to the now-old astdb format. > Also (and this is what interests me most), the sql > astdb could easily be clustered throughout several > servers (I'm looking for a master-master MySQL > 2-server cluster solution). > > Asterisk has brought up Realtime which is very > powerful but, correct me if I'm wrong, it still > requires astdb internally. In other words, if I call > Set(DB) in the dialplan then it will always be using > astdb regardless of realtime. > > Some projects like Callweaver have forked from > Asterisk 1.2 and replaced astdb with sqlite. > > I'm wondering if Asterisk has plans to allow the user > to choose the astdb backend: standard db1, sqlite, > MySQL (which I would use with nbcluster for my > clustering purposes), Postgresql with Slony-II, > PGcluster, etc. > > Or is it already possible? > > There has been some talk on this before: > http://lists.digium.com/pipermail/asterisk-dev/2004-December/007846.html > > Also, the func_odbc feature seems to be very powerful: > http://www.asteriskpbx.org/func_odbc > but: > 1) would there be potential issues with db handles on > a very busy asterisk system after a relatively long > run time? > 2) would there be a way to "map" the odbc function(s) > to the DB functions (Set(DB), read and write, DBdel, > etc) so that rewriting the whole dialplan would not be > necessary? (that's the whole point of defining a > different astdb "backend") > > If there are known > problems/issues/projects/alternatives then please let > me know. > > Thanks -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 79xx users/consultants, 7970G color in particular share information (was: Re: asterisk-users Digest, Vol 44, Issue 3)
-8859-1; format=flowed > > Sean Dennis ha scritto: > >> Sigma Networks wrote: > >> ... > >> My current questions are: > >> > >>1. How to remotely reboot 7970s. I have both web access and > SSH > >> access to the phones. The instructions I have for SSH are to > use > >> (1) user/pass (or whatever is in the confg) and then (2) > >> debug/debug. Surprisingly "reset" is not a valid command to > >> restart the phone. There doesn't appear to be a reset on the > web > >> page, maybe there's a hidden URL? > >>2. BusyLampField? > >>... > > We have about 200 79x1's running SIP w/ asterisk and we are very > pleased > > despite some of the non-standard things Cisco does. > > In answer to question 1 the only way we have found to reboot the > phone > > remotely is shutdown the port on the POE switch. This will drop > the > > PC's network as well if it is plugged into the phone. > > Question 2 I would like to know the answer to myself. I would be > > curious to know if it works with the SIP image in call manager. > > Same here. > > We have about 500 phones, from both 79x1 and 79x0 series; > I posted the same two questions twice some time ago but never > got an answer: I do reboot phones by power cycling them too, > while I've been able to use blf with sccp images only. > > Furthermore, XML Services on 7940/7960 seem to be broken > or at least to behave in different way than the one > described in the sdk documentation. > > I needed the reboot feature to implement extension mobility but > I wasn't able to find a clean way. Power cycling is not always > an usable method, as many phones are powered by the AC adaptor. > I think I will able to put my hands on an UCM6.1 box very soon > to try that out and eventually grab the xml profiles. > As soon as I get the info I'll surely post it on this ML and on > voip-info too. > > Alberto. -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SiP call generator
Is there a simple tool that I can use to script Asterisk generating lots of calls according to a peak traffic curve, with random variance within a specified percentage around that curve, to test a number of DIDs at which I terminate voice recordings to test the audio and call quality? Any that will also give me a report of the actual traffic connections? On Tue Feb 19 09:00:45 CST 2008 Atis Lezdins wrote: > On 2/19/08, Alex Balashov wrote: >> Or, you can write your own scripts to generate calls via the Manager >> API, or use Asterisk call files (see voip-info.org on this topic). >> >> But, all other things being equal, it is probably preferred to use some >> sort of testing framework of the sort mentioned below. > > The PBX Testing Framework i mentioned (and also developed) provides > call-generation trough call-files so all you have to do is code action > scripts (answer, talk for 3-10 minutes, transfer to other extension, > etc..) and call generation scripts (random agent call every 10-20 > seconds, and random customer call every 20-30 seconds), all in PHP > with some functions and objects to make interaction easy. > Atis > >> Atis Lezdins wrote: >>> On 2/18/08, Khaled Chehab wrote: >>>> >>>> >>>> I want to have a PC-based real-time VoIP bulk call generator (including >>>> both >>>> SIP signaling and RTP generation) >>>> >>>> for stress testing and precise analysis of the VoIP network equipment. >>>> >>>> >>>> >>>> Do any one knows a free program can do that . >>> If you want just simple calls, i suppose SIPP can do that. >>> http://sipp.sourceforge.net/ >>> >>> If you want to have those calls perform some actions (send DTMF, etc), >>> you can try to write your own scripts based on PBX Testing Framework. >>> http://ftp.iq-labs.net/pbx-test/pbx-test-0.1.0.tar.gz Currently it's >>> designed for testing queue-agents scenarios but i'm sure you can >>> adapt. >>> Atis >> Alex Balashov -- Alex Balashov -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is a "secure call"?
If Asterisk does indeed use SECUREDIAL or similar as distinct from
DIAL, then DIAL should wrap SECUREDIAL for calls to a party that are
secure. This would parallel HTTP "GET" (or "POST") which use the same
function entry for both secure and insecure connections, wrapping their
secure access inside generic access.
To continue the parallel, the dialstring should indicate whether
SIP/TLS (and otherwise for IAX) is to be used, which should allow the
DIAL function to determine whether to make a secure connection. To go
further, if SECUREDIAL is invoked on a dialstring which does not specify
a secure connection, that invocation should at least flag the insecure
connection attempt, or even fail with an exception.
I'm not sure that the SIP spec allows a request for an insecure
connection to be rejected with instructions for requesting a secure
call. But if it does, then the DIAL function should allow logic for
options on the retry, like just failing with exception report or a list
of dialstrings to retry. Or maybe just an extention to jump to with the
failure in a variable, for the dialplan/AGI/etc able to use that status
for logic on retry or fail.
In general, the closer the DIAL function works to familiar Web
retrieval functions, the more Web programming techniques will be
applicable to Asterisk programming.
On Wed, 2008-02-13 at 10:40 -0600,
[EMAIL PROTECTED] wrote:
> Date: Wed, 13 Feb 2008 15:22:10 +0100
> From: Johansson Olle E <[EMAIL PROTECTED]>
> Subject: [asterisk-users] What is a "secure call"?
> To: Asterisk Non-Commercial Discussion Users Mailing List -
>
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes
>
> Friends,
>
> The following mail was sent earlier to asterisk-dev and did not
> cause
> the amount of discussion I hoped it would.
> Now that we have a way to secure signalling in IAX2 and SIP in
> Asterisk svn trunk, we need to start working on
> the concept of a "secure call" - or does it really matter?
>
> In SIP, there's a specification for how I as a domain owner can
> request all calls to my domain to use
> SIP/TLS by using DNS NAPTR and SRV records. But how do I as a caller
> request a secure service?
> How do we place a secure call with DIAL? Do we need SECUREDIAL?
>
> Any ideas and thoughts on the subject are welcome!
>
> Regards,
> /Olle
>
> - Copy of earlier mail -
> (http://lists.digium.com/pipermail/asterisk-dev/2007-July/028377.html)
>
> To open a can of worms... :-)
>
> I'm involved in Phil Zimmerman's efforts to integrate Zrtp into
> Asterisk. At the same time we have code for SRTP that needs to
> be integrated.
>
> This means that we will add the concept of a "secure call" in
> Asterisk. At some point, I want to be able to build dialplans
> where I can manager security requirements on channels, like "this
> conference is protected and requires a secure channel".
>
> So, to make this easy, should we have a boolean flag and determine
> "this is a secure call according to Asterisk Community
> Security Standards" or how should we handle this? I think leaving
> it
> up to the admin is the proper way to go, but we
> also have several scenarios to consider
>
> 1. Encrypted signalling and media stream
> 1. Open signalling stream, key exchange in the open, encrypted media
> 2. Open signalling stream, secure key exchange, encrypted media
> 3. Secure signalling stream, un-encrypted media
>
> exten => _x.,n,gotoif(${ISSECURECALL(level6)} ? approved,1 :
> hangup,1)
>
> And to add to that, we have many different call scenarios.
>
> 1. Bridged call between two secure endpoints, Asterisk transcodes
> and
> have an unsecure media path
> 2. One-legged secure call between Asterisk and a phone (IVR)
> 3. SIP to ASterisk over IAX trunk to another Asterisk to SIP with
> SRTP/
> TLS and encrypted IAX - but open
> media path when going from SIP to IAX
>
> And yes, of course, this is not attempting to be a complete list at
> all.
>
> Can we simplify this and make it configurable? Do we want to?
>
> Can we implement the notion of a "trusted" PBX that we allow being
> in
> the middle and "untrusted" PBXs
> that we want to avoid or that changes the security property of a call.
>
> As I said to Phil: "A PBX is designed to be a man-in-the-middle
> attack."
>
> There's certainly room for discussion, brainstorming and wild ideas
> here.
>
> /O
>
>
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Re: [asterisk-users] Replacement for Allison
On Thu, 2008-01-24 at 13:51 -0500, Steve Prior wrote:
> Matthew Rubenstein wrote:
> > Is anyone else interested in creating new voices for Festival (the
> > voice synth bundled with Asterisk) that might not be as good as
> > Allison's recordings, but are better than the current Festival voices?
>
> If you try to do live voice synth for prompts you'll probably run into
> something I encountered - the background and speechbackground (I also
> use Lumenvox in addition to Cepstrel) only take a file. That means
> unless I'm missing something you can't have a TTS prompt that can be
> interrupted like a recorded file. The workaround of TTS to a file and
> then play the file sounds to me like it would introduce delays and
> besides it's ugly.
Since Asterisk has a preset collection of prompts, the voice synth can
be used to generate those files when installing Asterisk (or whenever).
The difference from Allison's prompts is that when a new app calls for a
new prompt, the synth can generate its new file instead of hiring
Allison to do it, and all the prompts sound consistent. And there are
ways to synthesize arbitrary new prompts at runtime just in time to be
picked up by the apps that can play only a file.
> That's why I posted a suggestion to the recently created
> asteriskideas.org that background and speechbackground be enhanced to
> take an app in addition to a simple file. If you agree, then please
> vote for it.
I think the flexibility you described is important anyway.
> Steve
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Re: [asterisk-users] Replacement for Allison
Is anyone else interested in creating new voices for Festival (the
voice synth bundled with Asterisk) that might not be as good as
Allison's recordings, but are better than the current Festival voices?
On Thu, 2008-01-24 at 12:00 -0600,
[EMAIL PROTECTED] wrote:
> Date: Thu, 24 Jan 2008 11:14:28 -0500
> From: Matt <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] Replacement for Allison
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Message-ID:
> <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="iso-8859-1"
>
> That worked... hrmm not that great... anyone know of any decent
> sounding
> recording of Allison for Asterisk?
>
> On Jan 23, 2008 11:26 PM, Andrew Joakimsen <[EMAIL PROTECTED]>
> wrote:
>
> > for x in *.g711u; do mv "$x" "${x%.g711u}.ulaw"; done
> >
> > On Jan 23, 2008 5:00 PM, Matt <[EMAIL PROTECTED]> wrote:
> > > Hi,
> > > Does anyone know what I need to do to get these:
> > > http://www.enicomms.com/cutglassivr/
> > >
> > > Sounds files to work? I've tried loading them, but they are
> completely
> > > silent (format mis-match maybe?). Specifically, when I try to
> enter
> > > voicemail, nothing plays... though it clearly tries.
> > >
> > > I'm looking for replacement sound files for the default Allison,
> as I
> > feel
> > > she is kind of breathy. I have heard other sound files on other
> > asterisk
> > > sounds, done by her, and they sound fine... are there "two"
> recorded
> > > versions of the prompts floating around?
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Re: [asterisk-users] Nightly tarballs, would you use them?
I'd be even more likely to use nightly (or other periodic snapshot, even weekly) .deb packages. Because then I could use APT to notify me and manage them. Especially if they included a changelog (which APT reports), even if that changelog were only names of files/modules touched since the last one. On Sat, 2008-01-19 at 12:00 -0600, [EMAIL PROTECTED] wrote: > Date: Sat, 19 Jan 2008 03:21:54 -0600 > From: Russell Bryant <[EMAIL PROTECTED]> > Subject: [asterisk-users] Nightly tarballs, would you use them? > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1 > > Greetings, > > During the past week, there have been some requests for nightly > tarballs to help > making testing new Asterisk code easier. There was some debate as to > whether > they would be useful. The reason that they may not be useful is > because you can > get equivalent access to new code just by accessing the subversion > repository > directly. However, for one reason or another, some people would > prefer to have > a tarball. > > If this was available, would you be interested in it? > > If you just want to say "yes or no" for the sake of the poll, fell > free to > respond to me off-list. However, also fell free to respond here if > you have > more verbose comments on the topic that you would like to share. > > -- > Russell Bryant -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discover Asterisk 1.4 :: Google Talk, XMPP and Jabber Integration!
On Sat, 2008-01-12 at 08:35 -0600, [EMAIL PROTECTED] wrote: > Date: Sat, 12 Jan 2008 11:02:17 +0100 > From: Johansson Olle E <[EMAIL PROTECTED]> > Subject: [asterisk-users] Discover Asterisk 1.4 :: Google Talk, XMPP > and Jabber Integration! > To: Asterisk List - Non-Commercial Discussion Users Mailing > > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes > > I've written a new article about Asterisk 1.4's Jabber integration. > Check it out at > http://www.voip-forum.com/asterisk/2008-01/xmpp/ > > /Olle [from http://www.voip-forum.com/asterisk/2008-01/xmpp/ ]: * Jabber presence support in the dialplan: By letting your Asterisk connect to a Jabber server by using a Jabber account, you can add buddies to that account and check the buddies presence in the Asterisk dialplan. This way, call routing decisions can be based on the status of Jabber accounts. (...) * Asterisk as a Jabber module: In a more advanced mode, Asterisk can register itself as a module to your Jabber server (as a Jabber component). This mode means better integration to Jabber, but requires more from the Jabber clients. [/from] Can an Asterisk server hold logins for multiple Japper accounts on a remote Jabber server, and carry multiple Jabber calls simultaneously the way it can carry multiple SIP (or IAX, or ZAP, etc) calls? If so, is each of those Jabber calls as lightweight as, say, each SIP call? If not, is there a way to increase the capacity of Asterisk to carry about as many Jabber calls as it can carry SIP calls? -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 79xx XML services
I Googled for CMXML_App_Guide.pdf . The first result was the voip-info wiki article "Asterisk phone cisco 79xx" at http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx . That article mentions CMXML_App_Guide.pdf in the "Company Telephone Directory" section, with a link to the "Asterisk Cisco 79XX XML Services" wiki article at http://www.voip-info.org/wiki/view/Asterisk+Cisco+79XX+XML +Services . Which in turn mentions that the relevant Cisco doc is called "Cisco IP Phone Services Application Development Notes (Cisco IP Phone XML Objects)". So I Googled for "Cisco IP Phone XML Objects" which turned up several results for a 2002 O'Reilly book, followed by the doc itself at http://www.cisco.com/univercd/cc/td/doc/product/voice/vpdd/cdd/5_0/5_0_1/ipphsv/ip503ch2.htm . Along the way there were several other docs and examples, including differences between SIP/SCCP version of the service. I'm interested to see how well the feature works on the 7970s with Asterisk instead of CallManager. Please keep me posted on your progress. On Fri, 2008-01-04 at 17:08 -0600, [EMAIL PROTECTED] wrote: > Date: Fri, 04 Jan 2008 13:41:31 -0800 > From: Edwin Lam <[EMAIL PROTECTED]> > Subject: [asterisk-users] Cisco 79xx XML services > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=UTF-8; format=flowed > > > hi guys. > > i'm writing some simple applications for the cisco 7970 > services button. i read the asterisk wiki and it mention > there's a CMXML_App_Guide.pdf file but there's nowhere > can i find a link for it. does anybody know where can > i find it? > > regards. > -- > Edwin Lam <[EMAIL PROTECTED]> -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 7970 CTLFile.tlv?
I've got in.atftpd running out of inetd: - /etc/inetd.conf tftpdgram udp waitnobody /usr/sbin/tcpd /usr/sbin/in.tftpd --logfile /tmp/atftpd.log --pidfile /tmp/atftpd.pid --tftpd-timeout 300 --retry-timeout 5 --mcast-port 1758 --mcast-addr 239.255.0.0-255 --maxthread 100 --verbose=5 --no-blksize /tftpboot --- But even when I run use a tftp client from a host on the inside network to retrieve the SEP.cnf.xml file successfully, the /tmp/atftpd.log file is never touched, nor is the /tmp/atftpd.pid ever created. Even if I (touch /tmp/atftpd.log; chown nobody.nogroup /tmp/atftpd.log) the status files are untouched. But I am getting the requested file. Also, what do I do to use an XmlDefault.cnf.xml file? Just rename the SEP.cnf.xml file to that? I also saw on the Web someone who had my problem with the 7970, but cryptically noted that they solved their problem which was wrong platform newline terminations. What chars does the 7970 need for its conf files newlines to be? On Fri, 2007-12-21 at 09:47 -0600, Jason Parker wrote: > You don't need the .tlv file. It's optional, and will be skipped if it cannot > be found. Your problem is elsewhere. I've found that the 7970s are very > finicky. I've never had luck with the SEP.cnf.xml - only > XmlDefault.cnf.xml (case may vary - check your tftp logs) > > Matthew Rubenstein wrote: > > I've got a Cisco 7970 that's not completing its network registration to > > Asterisk. The "Registering" message stays on the screen (with the moving > > time wheel). After a few minutes, the onscreen message flashes "Updating > > CTL" then "Loading...", then the status messages update with: > > > > No valid CAPF server > > File Not Found: CTLFile.tlv > > No CTL installed > > SEP.cnf.xml (where is the phone's MAC addr minus :s) > > > > before repeating the cycle (forever). > > > > Where can I get a CTLFile.tlv , or remove the requirement for it? Or is > > there another way to fix this problem? TIA. > > > > Asterisk v1.2.9.1-BRIstuffed-0.3.0-PRE-1q > > SCCP firmware > > Load File: TERM70.7-0-1-0s > > App Load ID: Jar70.2-9-0-117.sbn > > JVM Load ID: CVM70.2-0-0-112.sbn > > OS Load ID: cnu70.2-7-4-134.sbn > > Boot Load ID: 7970_64060118.bin > > -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 7970 CTLFile.tlv?
I did (/tftpboot/CTLFile.tlv), but the phone keeps complaining (no change). On Fri, 2007-12-21 at 10:25 -0500, Anciso, Roy wrote: > I believe you can create a blank file to keep the phone from > complaining. > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Matthew > Rubenstein > Sent: Friday, December 21, 2007 10:16 AM > To: Asterisk -Users > Subject: [asterisk-users] 7970 CTLFile.tlv? > > I've got a Cisco 7970 that's not completing its network > registration to > Asterisk. The "Registering" message stays on the screen (with the moving > time wheel). After a few minutes, the onscreen message flashes "Updating > CTL" then "Loading...", then the status messages update with: > > No valid CAPF server > File Not Found: CTLFile.tlv > No CTL installed > SEP.cnf.xml (where is the phone's MAC addr minus :s) > > before repeating the cycle (forever). > > Where can I get a CTLFile.tlv , or remove the requirement for > it? Or is > there another way to fix this problem? TIA. > > Asterisk v1.2.9.1-BRIstuffed-0.3.0-PRE-1q > SCCP firmware > Load File: TERM70.7-0-1-0s > App Load ID: Jar70.2-9-0-117.sbn > JVM Load ID: CVM70.2-0-0-112.sbn > OS Load ID: cnu70.2-7-4-134.sbn > Boot Load ID: 7970_64060118.bin -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 7970 CTLFile.tlv?
I've got a Cisco 7970 that's not completing its network registration to Asterisk. The "Registering" message stays on the screen (with the moving time wheel). After a few minutes, the onscreen message flashes "Updating CTL" then "Loading...", then the status messages update with: No valid CAPF server File Not Found: CTLFile.tlv No CTL installed SEP.cnf.xml (where is the phone's MAC addr minus :s) before repeating the cycle (forever). Where can I get a CTLFile.tlv , or remove the requirement for it? Or is there another way to fix this problem? TIA. Asterisk v1.2.9.1-BRIstuffed-0.3.0-PRE-1q SCCP firmware Load File: TERM70.7-0-1-0s App Load ID: Jar70.2-9-0-117.sbn JVM Load ID: CVM70.2-0-0-112.sbn OS Load ID: cnu70.2-7-4-134.sbn Boot Load ID: 7970_64060118.bin -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trixbox Phones Home
I just read on Slashdot (at http://yro.slashdot.org/article.pl?sid=07/12/16/43 ) that Trixbox "has been phoning home with statistics about their installations", as a Trixbox user exposed in "Trixbox Phones Home" at http://www.trixbox.org/forums/trixbox-forums/open-discussion/trixbox-phones-home . -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Pcengines Alix board
Other than the Alix board, what else is needed to make a working PC? On Mon, 2007-11-19 at 07:28 -0600, [EMAIL PROTECTED] wrote: > Date: Sun, 18 Nov 2007 22:14:15 +0100 > From: Giuseppe Barichello <[EMAIL PROTECTED]> > Subject: [asterisk-users] Asterisk on Pcengines Alix board > To: asterisk-users@lists.digium.com > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=US-ASCII > > Hi all, > > I have successfully compiled and installed Asterisk on an Alix board > (AMD Geode 500 Mhz + 256 Mb RAM) on top of Voyage linux (a Debian > variant). > I'm using it at home for a month. > > I wondered how much it could be loaded, so I tested it with pbx-test: > I could place up to 15 simultaneous SIP calls before it got no more > responsive. > > All in all a good, stable and cheap solution for home and home-office > environments. > > My 2 cents, > > Giuseppe -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7940G licensing with asterisk
On Mon, 2007-10-01 at 19:02 +0200, Olivier wrote: > Matthew, > > Did you keep any hardcopy of licensing terms (when downloading SIP > firmware) ? > This way we might double check if CCM license is mandatory to connect > a Cisco SIP phone to an Asterisk server. I haven't seen any such mandate, and didn't elicit one when I told Cisco I was using the firmware/phones with Asterisk instead of CallManager. I don't think there is one. You can look at the release notes for all the 7900 firmware available for download, including the version I got: http://www.cisco.com/en/US/products/hw/phones/ps379/prod_release_notes_list.html . > Beside that, Cisco SIP phones require menu localization files to come > from CCM. > Did you run into this ? > Is there anything special with these phones that make those > localization files to be downloaded (I know that's another topic, but > while we're at it ...) I have not completed the deployment of the phones, as I've had other priorities. I have not yet run into that problem, or heard of it before, but it might be lying in wait later in the process. I'd like to know whether it is indeed a problem in using the phones with Asterisk, and how to solve it if so. > Regards -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7940G licensing with asterisk
On Mon, 2007-10-01 at 11:44 -0500, Jason Parker wrote: > Matthew Rubenstein wrote: > > I just got SIP firmware images from Cisco for installation on > 7970G. > The way I understand it, that $15 doesn't actually even give you the > right to > use the SIP firmware. It only gives you the right to "access" the > download area. > > The whole model is silly, at best. When I explained to each of the account reseller and the Cisco support that I was going to use the SIP firmware to connect to Asterisk, not CallManager, they each told me only that Cisco wouldn't support (trouble tickets and other tech support time) the system using Asterisk, though they did explicitly assure me (as does the documentation) that since the SIP firmware is RFC-compliant, it would work with any RFC-compliant server, not just CallManager (and so would work with SIP RFC-complaint Asterisk). It's a giant game of CYA. I spent hours getting my $15 worth from the SIP download. I'm surprised a bitter backlash hasn't made these SIP images widely available for download around the Web. I think they might have the serial# of the phone they're registered to when the account is created, and of course the contract states otherwise, but I'd still expect Cisco's deliberately difficult process hasn't created enemies who'd do it anyway. Maybe there are just so few people using it this way that none have materialized (yet). So I guess Cisco's PITA plan is working. -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7940G licensing with asterisk
I just got SIP firmware images from Cisco for installation on 7970G. Cisco requires you buy a SmartNet account (about $15, no other dependencies apply) that entitles you to download a SIP firmware image file from their protected support website. The 7970G now needs a different image than the other 79xx phones, but the same rules apply to all of them. Those rules do not require any other license or other restriction, once you have legitimately obtained and installed the firmware on the phone, to use the phones with Asterisk (or any other 3rd party system). Of course, to use the phones with Cisco's CallManager product, you must have a licensed copy of the CallManager product, with all the other restrictions and fees that come with it. FWIW, the procedure of buying that SIP image from Cisco was a nightmare. I had to buy the SmartNet account from a reseller which did nothing to ensure that I completed the download transaction that was the stated purpose (as they described it to me) of buying the license. Then navigating to the license I needed, among the many versions and revisions, was confusing and opaque. The SmartNet account took days to send to me, and didn't work for the required access when it arrived. Cisco consumed an entire workweek to deliver the license that didn't unlock the website, then of course ignored requests for support through the weekend (into which their late delivery forced my request to be made). When I finally got Cisco to respond, they did deliver a knowledgeable and honest support tech who stuck with me until I had everything I needed to proceed. Though every stated "maximum" turnaround time for every phase in the process was exceeded, sometimes by many multiples. But since the image can be used only with a Cisco phone, which must (ultimately) be bought from Cisco, the kafkaesque procedure is intolerable. The image should be a one-click download that charges your credit card and comes with a SmartNet account, if they absolutely must charge the $15. In a sane world, the SIP image wouldn't have any restrictions, a free download that people could just email each other (or its URL), because its distribution would market Cisco phones. But probably Cisco knows that the SIP image lets (free) Asterisk compete with its proprietary CallManager, so they make it both a revenue source, and as complicated as possible. On Mon, 2007-10-01 at 09:43 -0500, [EMAIL PROTECTED] wrote: > Message: 18 > Date: Mon, 1 Oct 2007 10:21:34 -0400 > From: "Glenn Cobb" <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Cisco 7940G licensing with asterisk > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="US-ASCII" > > In trying to verify licensing requirements I called Tech-Data and > spoke to > the Cisco licensing reps there (my company is set up as a reseller > through > Tech-Data) and was informed by them that a license for Cisco VoIP > phones is > only required if connecting it to a Call Manager or any other Cisco > voice > technology solution such as a Cisco router. If you are connecting a > Cisco > phone to any other pbx they consider it a "third party solution" and > licensing requirements for that vendor are your responsibility. > > Glenn -- (C) Matthew Rubenstein ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX Java Softphone?
Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DECT SIP phones
On Fri, 2007-09-14 at 12:00 -0500, [EMAIL PROTECTED] wrote: > Date: Fri, 14 Sep 2007 09:32:35 -0500 > From: Tilghman Lesher <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] DECT SIP phones > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > On Thursday 13 September 2007 19:05:51 Stephen Bosch wrote: > > I'm looking for a SIP DECT (cordless) phone for North American > > installations. I've heard only of the Siemens Gigaset S450/C450 > phones. > > Apparently these aren't sold for use in NAm, even though they're > > supposed to be legal (in the United States, anyway). > > > > On top of that, I understand they have some annoying issues anyway. > > > > Can anyone suggest a solid alternative DECT SIP phone that is > available > > in North America? > > I don't know how solid you would consider them, but I have repurposed > the > ATS X10001P phones that are sold for use with Lingo into phones that > can > be used with Asterisk. At $70US, I suspect they are the least > expensive > SIP DECT phones available. Wal-Mart sells the ATS X10001P for $55, and claims it has a "fax port": http://www.walmart.com/catalog/product.do?dest=97&product_id=6457851&sourceid=1503142050 . Is there a way to fax with these phones without Lingo? How does Lingo do it (over the phone's Internet connection), if Asterisk can't? > http://asterisk.drunkcoder.com/hacks/ats-config/ Your server seems very slow, often timing out. > -- > Tilghman > -- (C) Matthew Rubenstein ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 79xx XML Apps (was: Re: Cisco Directory Format)
Do you know where to find clear developers' guides (with some examples) for developing apps that run *on* Cisco 79xx phones (especially the 7970)? Examples that can run against Asterisk (not CallManager) with SIP firmware (not SCCP), and/or LDAP directories (or other open servers) would be best. On Sat, 2007-09-01 at 12:00 -0500, [EMAIL PROTECTED] wrote: > Date: Sat, 1 Sep 2007 12:14:49 -0400 > From: "Time Bandit" <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Cisco Directory Format > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1 > > > A little off topic (sorry..:) ) but anyone know what format Cisco > phones > > use for their contact dirctories. I want to set up my contact lists > on > > the phone, and cannot seem to get any info on it. I am working with > a > > 7970 on Asterisk 1.4.8. > 7940 and 7960 use this format of XML file (probably the same on 7970) > > > Employee directory > Open Source Rock > > Employee A > 7001 > > > Employee B > 7002 > > > > Check also Open 79XX XML Directory : > http://web.csma.biz/apps/xml_xmldir.php > > hope that help > -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7970G App Development?
Do you know where I can find docs for developing apps that run locally on a Cisco 7970G IP phone (with SIP firmware installed)? Apps that use the phone's display, keys, and other local functions, as well as call init/control, and other network features, including looking up directory info in, say, an LDAP server? All development using Asterisk instead of CallManager services, of course. -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stereo Conferences?
Do any softphones run the HD codec? What exactly is the HD codec technically called, and is there any info about its codec running inside Asterisk? On Mon, 2007-08-27 at 08:47 -0400, C F wrote: > although not stereo i believe its the closest you will get if the > codec is supported by asterisk. polycom has now HD codec > > On 8/27/07, Matthew Rubenstein <[EMAIL PROTECTED]> wrote: > > Are there any speakerphones or other conferencing HW phones that play > > the audio in stereo? Either their own speakers, or jacks for an amp with > > room speakers? Is there any way for Asterisk to deliver call legs with > > stereo channels in the RTP stream? > > > > If not, is it possible for Asterisk to keep 2 separate calls, or pairs > > of legs in a conference call, synced exactly enough (including traveling > > over the Net between the same 2 IP#s) for them to arrive as a stereo > > pair at the endpoint? > > -- > > > > (C) Matthew Rubenstein > > > > > > ___ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Stereo Conferences?
Are there any speakerphones or other conferencing HW phones that play the audio in stereo? Either their own speakers, or jacks for an amp with room speakers? Is there any way for Asterisk to deliver call legs with stereo channels in the RTP stream? If not, is it possible for Asterisk to keep 2 separate calls, or pairs of legs in a conference call, synced exactly enough (including traveling over the Net between the same 2 IP#s) for them to arrive as a stereo pair at the endpoint? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] xPL and Asterisk?
On Fri, 2007-08-24 at 03:44 -0500, [EMAIL PROTECTED] wrote: > Message: 20 > Date: Thu, 23 Aug 2007 23:13:55 -0500 > From: Jay Milk <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] xPL and Asterisk? > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Matthew Rubenstein wrote: > > I tried asking in another thread this week, but I'm not sure > people saw > > the actual subject of the question. Does anyone know where to find > > documentation of xPL, the home automation interface? Specifically > for > > integrating it with Asterisk. xPL is part of Trixbox, so it's being > > used, but where is some expertise for using it without Trixbox? > > > http://www.google.com/search?q=xpl+home+automation > > 1st and 3rd results. I actually mentioned the explicit Google search URL in my previous message to the list. But I also mentioned that I prefer the list's experience in actual use of xPL with Asterisk. I'm looking for specific xPL/Asterisk docs that Asterisk people have tested. The community is a source of best practices, which is what I'm looking for. Like insight into whether to use the xPLhub for Linux that's available, or whether there's a different way to go. -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] xPL and Asterisk?
I tried asking in another thread this week, but I'm not sure people saw the actual subject of the question. Does anyone know where to find documentation of xPL, the home automation interface? Specifically for integrating it with Asterisk. xPL is part of Trixbox, so it's being used, but where is some expertise for using it without Trixbox? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Home Automation (was: Re: 99 bottles of beer)
On Wed, 2007-08-22 at 08:50 -0500, [EMAIL PROTECTED] wrote: > Date: Tue, 21 Aug 2007 21:01:50 -0400 > From: "David Cook" <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] 99 bottles of beer > To: > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > On 8/21/07, Steve Edwards <[EMAIL PROTECTED]> wrote: > > > > > > "To control the tv in this room, press 1. To control a tv in > another > > > room, press 2. To control the outside lights, press 3. To control > the > > > sprinklers, press 4, ..." > > > > > > > Before this thread I already had a Firecracker on the server, a fair > assortment of lights and the sprinklers are on an X10Pro Irrigation > Controller. > > > > Damn, now I'm gonna be up all night. Isn't this kind of Asterisk interface to home automation what the xPL package in Trixbox is supposed to offer? Is there a source for clear, concise, *tested* guides and instructions for Asterisk/xPL home automation somewhere other than just a needle in the http://www.google.com/search?q=xpl+%22home+automation%22+asterisk haystack? Or maybe there's a better interface than xPL. -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-biz] Skype Outage Leaves Millions Speechless
On Fri, 2007-08-17 at 18:22 +0200, Trixter aka Bret McDanel wrote: > On 8/17/07, Aleks Clark <[EMAIL PROTECTED]> wrote: > > Actually, the crazy p2p connections actually reinforce their algorithm > > story. If their p2p algorithms have flaked out, it could cause all sorts of > > trouble. OTOH, I don't think they'd run logins over p2p > > > > given the press from days past about how someone cracked the algorithm > and could write their own client that ebay cant control, almost makes > you wonder if it was an auto-update gone awry to try to change the > algorithm. > > I dont know what the default setting is, but do know that skype can be > set to auto-update itself, which means that some may have been > affected while others werent for that reason alone. > > I am certain though that skype wouldnt admit if it was this, and its > likely that any front line people at skype wouldnt know one way or the > other for sure what is broke. Imagine if the world's largest online marketplace operated the world's largest alternative (and one of the largest in general) telco and an unregulated global online banking monopoly. And the telco suddenly went down, unexplained, for hours or days. That sounds like a serious threat to global economy and security, right? eBay is that marketplace, owns Skype, that telco, owns PayPal, that bank. This outage should be screaming from the headlines. As those three essential services become essential to more people around the world, they need to become reliable. This outage is a serious warning for future dependence on those connected services. If the media can't even report it, how can we expect anyone to do anything to fix or mitigate it? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Client-negotiated Codec Instead of Transcoding?
Is there a way for voice media clients (like SIP phones and POTS/PSTN phones) that connect their call legs to Asterisk to negotiate a common codec that they both use at their end, so Asterisk doesn't have to transcode? Asterisk would know which codecs each client can use, and which each prefers, then find the one they each have in common so the fewest legs need Asterisk to transcode to their "odd man out" codec. -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] text2wave Voices Improvements?
On Sun, 2007-08-05 at 20:32 -0500, [EMAIL PROTECTED] wrote: > Date: Sun, 5 Aug 2007 19:08:25 -0300 > From: Jo?o Paulo Vanzuita <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] text2wave Voices Improvements? > To: asterisk-users@lists.digium.com > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=US-ASCII > > On Sat, 04 Aug 2007 19:52:21 -0400 > Matthew Rubenstein <[EMAIL PROTECTED]> wrote: > > > I currently have an AGI that calls the Festival text2wave app > to write > > a wav file that my dialplan plays into a call with the Background() > > command. But the voice sounds terrible: like SAM, the 1980s 6502 > voice > > synthesizer. I tried to slow it down by calling (text2wav -eval > > "(Parameter.set 'Duration_Stretch 1.4)" -scale 2.0 [...]), but it > still > > sounds like it's talking while sucking down a strawful of spaghetti. > How > > do I install a different voice, to speak basically simple emails? > I'm > > (APT) installing on Debian 3.1/Sarge, Asterisk 1.4.x . > > works fine to me installing "festvox-kallpc16k" for speaker > apt-get install festvox-kallpc16k festival I apt-get install'ed that package, and rablpc16k and kdlpc16k . But though it rab and kal voices work, kdl does not: #text2wave -o say.wav say.txt -eval "(voice_kdl_diphone) (Parameter.set 'Duration_Stretch 1.4)" -scale 2.0 SIOD ERROR: unbound variable : voice_kdl_diphone They all sound like a 1980s synth (which wasn't so bad, if you were familiar with the burbly voice). I'm using the first 2 sentences of (man man) as my test string: "man is the system’s manual pager. Each page argument given to man is normally the name of a program, utility or function." Maybe there's some kind of hint characters I can insert to make it sound better? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] text2wave Voices Improvements?
I currently have an AGI that calls the Festival text2wave app to write a wav file that my dialplan plays into a call with the Background() command. But the voice sounds terrible: like SAM, the 1980s 6502 voice synthesizer. I tried to slow it down by calling (text2wav -eval "(Parameter.set 'Duration_Stretch 1.4)" -scale 2.0 [...]), but it still sounds like it's talking while sucking down a strawful of spaghetti. How do I install a different voice, to speak basically simple emails? I'm (APT) installing on Debian 3.1/Sarge, Asterisk 1.4.x . Also, is there a way to call Background or some other Asterisk command to take the WAV data from a pipe to a running text2wav process, rather than writing a file with text2wave and then reading it (and then deleting it) in the dialplan/AGI? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lightweight IAX balancer
On Mon, 2007-07-30 at 07:01 -0500, [EMAIL PROTECTED] wrote: > Date: Mon, 30 Jul 2007 12:19:13 +0100 (BST) > From: Stanis?aw Pitucha <[EMAIL PROTECTED]> > Subject: [asterisk-users] Lightweight IAX balancer > To: asterisk-users@lists.digium.com > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=utf-8 > > Hi list > > I've written a tool that works as a lightweight (standalone - no > asterisk) balancer for IAX servers. It's in early development now, but > seems to be stable enough and handles couple hundred simultaneous > calls with not much latency (SIPp + asterisks tested). > It's configurable by listing servers' IPs in iaxproxy-servers file > loaded at startup and will keep track of load on each machine. > It does balancing not per IAX connection, but per call - rewriting > call numbers and keeping track of connection status. It's going to be > optimized for speed - doesn't do any other modification or audiostream > translation - only message passing. > > If someone's interested -- code + short doc is available at > http://www.gradwell.com/tmp/iax_proxy.tar.gz > > Development will continue - any opinions / comments / contributions > are appreciated. That SW looks like a valuable service. What are the chances you could code it into a module for OpenSER, so OpenSER could deliver both SIP and IAX routing/proxying, without having to rewrite all common parts of OpenSER to deliver its services to SIP? Also, OpenSER/IAX would make calls with mixed IAX/SIP legs easier to manage. And there's probably lots of performance optimization - not to mention deployment optimization. > Stanis?aw Pitucha > Gradwell Dot Com -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Corporate Feedback to OSS (was: Re: Mushtaq Ahmed is out of the office.)
Until they rip off your IP, or just use all the public contributions in combination with their better funded proprietary operation, without contributing anything themselves, or even admitting they're using the tech that could use the corporate boost. FWIW, 3Com is not an "Asterisk vendor". On Tue, 2007-07-17 at 10:49 +1000, Paul Hales wrote: > We have found that working WITH other Asterisk vendors is much more > pleasant than working against them - especially when you all run into > each other at a trade show. > > PaulH > > > On Sat, 2007-07-07 at 11:04 -0400, Matthew Rubenstein wrote: > > On Sat, 2007-07-07 at 08:39 -0500, > > [EMAIL PROTECTED] wrote: > > > Date: Fri, 06 Jul 2007 12:02:53 -0600 > > > From: Stephen Bosch <[EMAIL PROTECTED]> > > > Subject: Re: [asterisk-users] Mushtaq Ahmed is out of the office. > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > > > > Message-ID: <[EMAIL PROTECTED]> > > > Content-Type: text/plain; charset=ISO-8859-1 > > > > > > Wayne wrote: > > > > I was wondering where 3Com were getting all the new ideas from for > > > their > > > > phone system ;-p > > > > > > > > Cats out of the bag now I guess :) > > > > > > The price of open source is that the commercial outfits are free to > > > rip > > > off ideas without paying for them. > > > > > > But hey -- competition is good, right? > > > > Competition is good, one benefit of OSS pressure on > > commercial/proprietary competitors to improve their products which lead > > investment. > > > > Cooperation is also good. Public knowledge that corporations are in the > > community helps us know where to look for GPL software they secretly > > use, or just how they get some valuable ideas from which they profit > > (profit from us, usually). So it's easier to convince them to explicitly > > feed back into the OSS. Either just user feedback, or actual investment > > in testing, further development, or even GPL'ing their own proprietary > > tech into the community. > > > > So now it's time that 3Com hears from us, and we hear back, that we're > > all "coopeting" together. If they don't explicitly contribute soon, that > > bad community attitude will be a clue for some examination of their > > products for included GPL code and GPL violations, or just some bad > > press for being merely "takers" with their $billion budgets. > > > > > > > -Stephen- > > > > > > ___ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Corporate Feedback to OSS (was: Re: Mushtaq Ahmed is out of the office.)
True. But I think that fuzzy distinction is also relevant to the fuzzy process. I'm not talking about suing or fighting anyone, with actual evidence suitable to that kind of action. I'm just talking about clues for looking for actual evidence of actual actions. Besides, Mushtaq Ahmed's recent list posting seems like he's testing Asterisk/SIP while at work at 3Com, though it's not conclusive: http://archives.free.net.ph/message/20070329.164411.3b3da82d.en.html , and appears to have helped patent a PSTN/ethernet "conference call system" for 3Com: http://uspto.gov/web/patents/patog/week31/OG/html/1309-1/US07085364-20060801.html . Certainly one to watch, as he's watching us and Asterisk. On Sat, 2007-07-07 at 08:32 -0700, Tom Lynn wrote: > On the other hand, the guy could just be using his work e-mail for > personal interests. > > On 7/7/07, Matthew Rubenstein <[EMAIL PROTECTED]> wrote: > On Sat, 2007-07-07 at 08:39 -0500, > [EMAIL PROTECTED] wrote: > > Date: Fri, 06 Jul 2007 12:02:53 -0600 > > From: Stephen Bosch <[EMAIL PROTECTED]> > > Subject: Re: [asterisk-users] Mushtaq Ahmed is out of the > office. > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > > Message-ID: <[EMAIL PROTECTED]> > > Content-Type: text/plain; charset=ISO-8859-1 > > > > Wayne wrote: > > > I was wondering where 3Com were getting all the new ideas > from for > > their > > > phone system ;-p > > > > > > Cats out of the bag now I guess :) > > > > The price of open source is that the commercial outfits are > free to > > rip > > off ideas without paying for them. > > > > But hey -- competition is good, right? > > Competition is good, one benefit of OSS pressure on > commercial/proprietary competitors to improve their products > which lead > investment. > > Cooperation is also good. Public knowledge that > corporations are in the > community helps us know where to look for GPL software they > secretly > use, or just how they get some valuable ideas from which they > profit > (profit from us, usually). So it's easier to convince them to > explicitly > feed back into the OSS. Either just user feedback, or actual > investment > in testing, further development, or even GPL'ing their own > proprietary > tech into the community. > > So now it's time that 3Com hears from us, and we hear > back, that we're > all "coopeting" together. If they don't explicitly contribute > soon, that > bad community attitude will be a clue for some examination of > their > products for included GPL code and GPL violations, or just > some bad > press for being merely "takers" with their $billion budgets. > > > > -Stephen- > > > ___ > --Bandwidth and Colocation Provided by > http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Corporate Feedback to OSS (was: Re: Mushtaq Ahmed is out of the office.)
On Sat, 2007-07-07 at 08:39 -0500, [EMAIL PROTECTED] wrote: > Date: Fri, 06 Jul 2007 12:02:53 -0600 > From: Stephen Bosch <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Mushtaq Ahmed is out of the office. > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1 > > Wayne wrote: > > I was wondering where 3Com were getting all the new ideas from for > their > > phone system ;-p > > > > Cats out of the bag now I guess :) > > The price of open source is that the commercial outfits are free to > rip > off ideas without paying for them. > > But hey -- competition is good, right? Competition is good, one benefit of OSS pressure on commercial/proprietary competitors to improve their products which lead investment. Cooperation is also good. Public knowledge that corporations are in the community helps us know where to look for GPL software they secretly use, or just how they get some valuable ideas from which they profit (profit from us, usually). So it's easier to convince them to explicitly feed back into the OSS. Either just user feedback, or actual investment in testing, further development, or even GPL'ing their own proprietary tech into the community. So now it's time that 3Com hears from us, and we hear back, that we're all "coopeting" together. If they don't explicitly contribute soon, that bad community attitude will be a clue for some examination of their products for included GPL code and GPL violations, or just some bad press for being merely "takers" with their $billion budgets. > -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gigabit SIP Phones
Actually, uncompressed HDMI is 10.2Gbps. I don't think there are any 1080p IP phones yet, but there could be VoIP HD TVs coming. While there are some MPEG-2 apps with each stream consuming over 20Mbps ( http://en.wikipedia.org/wiki/MPEG-2#Profiles_and_Levels ), there could indeed be IP phone apps I've never thought of which could use >100Mbps, especially for multiple simul streams. There certainly will be eventually. On Wed, 2007-06-13 at 10:15 +0530, Vamsi Pottangi wrote: > >>> Also, are there any IP phones that run apps other than > telephony, like > >>> video, which could use more than 100Mb, even if just while > switching > >>> streams? > > Video of 100Mb/s? ;-) HDTV doesn't consume more than 20Mb/s, Gige is > an overkill for IP Phone. Though it is used for switching, I assume it > is a 1 in 100 use. > > Thanks, > ~Vamsi > > On 6/13/07, Matthew Rubenstein <[EMAIL PROTECTED]> wrote: > On Tue, 2007-06-12 at 16:44 -0700, > [EMAIL PROTECTED] wrote: > > Date: Tue, 12 Jun 2007 17:56:34 -0500 > > From: Darrick Hartman <[EMAIL PROTECTED]> > > Subject: Re: [asterisk-users] Gigabit SIP Phones > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > > Message-ID: <[EMAIL PROTECTED]> > > Content-Type: text/plain; charset=UTF-8; format=flowed > > > > Andrew Latham wrote: > > > Oliver > > > > > > The thing you missed about Gigabit enabled SIP hardphones > is the > > > demand for them. > > > > Not true. I can think of several places where I have or > would like > > to > > install phones where the end users currently have Gigabit > ethernet > > feeds > > to workstations. Specifically if you are using a > high-overhead > > system > > like Quickbooks Point of Sale and need a phone at the same > location, > > the > > end users will notice a significant performance hit by > dropping them > > down to 100Mbit. > > > > It's not so much that the phone needs Gig, it's that the > pass thru > > connection needs gig. > >And if you've got GigE installed, not 10/100Mb, and > your LAN doesn't > have a switch that can handle a phone's lower bitrate without > bringing > down the whole LAN's rate. > >Also, are there any IP phones that run apps other than > telephony, like > video, which could use more than 100Mb, even if just while > switching > streams? > > > > > Andrew > > > > > > On 6/12/07, Olivier < [EMAIL PROTECTED]> wrote: > > >> Hello, > > >> > > >> Beside Cisco 79x1-GE, I'm not aware of any Gigabit SIP > Phone. > > >> Did I miss something ? > > >> > > >> Regards > > >> > > >> ___ > > >> --Bandwidth and Colocation provided by Easynews.com -- > > >> > > >> asterisk-users mailing list > > >> To UNSUBSCRIBE or update options visit: > > >> > > >> http://lists.digium.com/mailman/listinfo/asterisk-users > > >> > > >> > > > > > > > > > > > > -- > > Darrick Hartman > -- > > (C) Matthew Rubenstein > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Gigabit SIP Phones
On Tue, 2007-06-12 at 16:44 -0700, [EMAIL PROTECTED] wrote: > Date: Tue, 12 Jun 2007 17:56:34 -0500 > From: Darrick Hartman <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Gigabit SIP Phones > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=UTF-8; format=flowed > > Andrew Latham wrote: > > Oliver > > > > The thing you missed about Gigabit enabled SIP hardphones is the > > demand for them. > > Not true. I can think of several places where I have or would like > to > install phones where the end users currently have Gigabit ethernet > feeds > to workstations. Specifically if you are using a high-overhead > system > like Quickbooks Point of Sale and need a phone at the same location, > the > end users will notice a significant performance hit by dropping them > down to 100Mbit. > > It's not so much that the phone needs Gig, it's that the pass thru > connection needs gig. And if you've got GigE installed, not 10/100Mb, and your LAN doesn't have a switch that can handle a phone's lower bitrate without bringing down the whole LAN's rate. Also, are there any IP phones that run apps other than telephony, like video, which could use more than 100Mb, even if just while switching streams? > > Andrew > > > > On 6/12/07, Olivier <[EMAIL PROTECTED]> wrote: > >> Hello, > >> > >> Beside Cisco 79x1-GE, I'm not aware of any Gigabit SIP Phone. > >> Did I miss something ? > >> > >> Regards > >> > >> ___ > >> --Bandwidth and Colocation provided by Easynews.com -- > >> > >> asterisk-users mailing list > >> To UNSUBSCRIBE or update options visit: > >> > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > >> > > > > > > > -- > Darrick Hartman -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Searchable List Archives?
I'd like to be able to search the list archives when I'm reading someone's message to put what they say in context based on what they've said, and what others have said in conversation with them, in the past. It would help me figure out whether to trust some submitters on some issues, and just learn more from the community's collective/cumulative research and discussion. Is there list server Web SW that lets me look at a message in the archives, then click on it to get every message (*across all months*) sent by that author, then every message in the thread (by Message-ID and same/similar subject)? Based on searches by regexp in each message field, including Body. Maybe Digium could upgrade the list SW, or let me do it for them. Or I could set it up at my website, then import the list archive data and parse it into my DB for a searchable mirror. Does the SW with those features exist already, or do I have to write it? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
OT: "The Ignorance of Crowds" (was: [asterisk-users] OT Slightly: )
I see what Dean means about how Digium/Asterisk might have struck a balance between "the cathedral and the bazaar" antipodes of the SW development world. Nicholas Carr's "The Ignorance of Crowds" finally states his "politics" when it says "When you move from the bazaar to the cathedral, it’s best to leave your democratic ideals behind." But treating open/closed source/projects as a pure dichotomy of two extremes of openness is a purely ideological exercise: and one that favors the cathedral, the very institution of ideology rather than practice. There are many degrees of openness, even just in the categories of the source code and of the project management. There are degrees of openness in the readability, writeability and executeability in each of those categories, to extend a metaphor. And there are other abilities, like redistribution, documentation, training, etc, which can be open to varying degrees. And any project can mix practically any openness degree in practically each of those abilities, for a vast combinatoric range. And calling the bazaar "democracy" is to misunderstand, and probably treat with contempt, both democracy and the *anarchy* of the market. Even the article's example that Dean highlighted, Wikipedia, shows no real "democracy", even the pure Athenian version that few Americans (except maybe some Californians) would recognize. Without actual rule by all of its contributors and readers, but rather primary rule by many policies determined and (often) enforced by people selected by autocrats (however benevolent), it's no democracy, but rather a collegiocracy or something else with a new name. Digium/Asterisk is an interesting example. For example, the community has so far accepted the proprietary ownership of code contributed to Digium, but a tension in source code openness lies in that degree in that category. The recent decision to stop new development of 1.2 in favor of 1.4 has just begun to enter the community consciousness, but the state of 1.4 when the 1.2 deadline comes will probably demonstrate limits of the project's openness to at least some committed 1.2 users/developers. Digium's "Asterisk" trademark hasn't yet become an issue, AFAIK, but a confusingly named fork, or just competing app from a different codebase with a very similar name could make all the current "Aster*" names into precedent damaging to the trademark, if not the mark itself. Digium is a corporation: an autocracy, not a democracy. It offers no data to judge democracy in its cathedral ruling its bazaar. And there are no deductively "identical but for one" versions of Digium run instead as a democracy to which to directly compare. Cathedral/bazaar is not a binary choice. They're more like antitheses that projects combine into a synthesized community model somewhere in the sphere of control combinations. It's too early to judge Digium's Asterisk success, let alone use it as a benchmark to calibrate cathedral/bazaar combinations. At least we have some terms in which we can model these complex behaviors and try to compare them. I don't think either the bazaar or the cathedral is in any way limited by, or alien to, "democratic ideals". A much more wise politics comes from Yogi Berra, who said "there is no difference between theory and practice - in theory". Let's keep trying the best way of running each job, and judge from the results when we've got examples of each. We can call them names when they've demonstrated what precedents they're actually like, and who likes them. What do you think? On Fri, 2007-06-01 at 05:42 -0700, [EMAIL PROTECTED] wrote: > Date: Fri, 1 Jun 2007 08:42:48 -0400 > From: "Dean Collins" <[EMAIL PROTECTED]> > Subject: [asterisk-users] OT Slightly: > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > Interesting article in this months S&B > http://www.strategy-business.com/press/enewsarticle/enews053107?pg=0 > > > > Written by Nicholas Carr - The Ignorance of Crowds "The open source > model can play an important role in innovation, but know its > limitations". > > > > At first pass I dissed it and was about to write back to Art Kleiner > the > editor about how BAH should stick to what it knows and was about to > provide references on the Asterisk development as a shining example of > Open Source at it's best..but when you read it the second or third > time on the 3rd and 4th page it starts to get interesting. > > > > Maybe the implementation Digium/Asterisk has struck is a per
RE: [asterisk-users] Zonbu
How much does a Patton NanoServ 607x cost? Their page has no price, an
inactive "Ordering" tab, Google doesn't have ("nanoserv 6070" price) in
its index (except a couple unresponsive del.ic.ious pages). PingTel
announce a SIPxNano based on it, for "under $1000" in 2006Q3:
http://www.pingtel.com/page.php?id=70&view=117 . Is there pricing for
just the HW without whatever bundled SW or service these telcos are
bundling/subsidizing it with?
On Sun, 2007-05-27 at 19:51 -0700,
[EMAIL PROTECTED] wrote:
> Date: Sun, 27 May 2007 23:18:26 -0300
> From: "Gustavo Cordeiro" <[EMAIL PROTECTED]>
> Subject: RE: [asterisk-users] Zonbu
> To: asterisk-users@lists.digium.com
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset=iso-8859-1; format=flowed
>
>
> $99,00 for one box, but you need a subscription plan...
>
> "Zonbu is $99 with a two-year subscription plan. With month to month
> plan,
> Zonbu is $249."
>
>
> Sds,
> Gustavo
>
> >From: "Nabeel Jafferali" <[EMAIL PROTECTED]>
> >Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial
> >Discussion
> >To: "'Asterisk Users Mailing List - Non-Commercial
> >Discussion'"
> >Subject: RE: [asterisk-users] Zonbu
> >Date: Sun, 27 May 2007 17:35:20 -0400
> >
> >Looks like a rebadged Patton 6075 to me:
> >
> >http://www.patton.com/products/pe_products.asp?category=337
> >
> >Nabeel
> >
> > > -Original Message-
> > > From:
> [EMAIL PROTECTED] [mailto:asterisk-users-
> > > [EMAIL PROTECTED] On Behalf Of Dean Collins
> > > Sent: May 27, 2007 11:53 AM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: [asterisk-users] Zonbu
> > >
> > > I just came across www.Zonbu.com <http://www.zonbu.com/> it's a
> > > fanless box about the size of a paperback book. It has no hard
> drive
> > > but runs it's Linux OS on a flash card - relying on document
> storage
> > > from an online service (rebadged Amazon S3).
> > >
> > > http://deancollinsblog.blogspot.com/2007/05/zonbu-net-pc.html
> > >
> > >
> > >
> > >
> > >
> > > I wonder who's going to be the first to hack an asterisk server
> onto
> > > this thing?
> > >
> > > At $99 it's a hell of an option for a fanless Asterisk server.
> > >
> > >
> > >
> > > Regards,
> > >
> > > Dean Collins
> > > [EMAIL PROTECTED]
> > > +1-212-203-4357 Ph
> > > +61-2-9016-5642 (Sydney in-dial).
> > >
> > > Call Button
> <http://click.mexuar.com/webuser/click/7/userurl/Cognation>
> > > <http://click.mexuar.com/webuser/nojs/7/userurl/Cognation>
--
(C) Matthew Rubenstein
___
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Re: [asterisk-users] PSP Voip
What was the content of the message you sent? And what is the deal with these messages the list delivers "scrubbed" of their content? Maybe the listbot can't handle "multipart/alternative" MIME messages. On Thu, 2007-05-24 at 05:52 -0700, [EMAIL PROTECTED] wrote: > Date: Thu, 24 May 2007 08:50:58 -0400 > From: "Dean Collins" <[EMAIL PROTECTED]> > Subject: [asterisk-users] PSP Voip > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > Skipped content of type multipart/alternative-- next part > -- > A non-text attachment was scrubbed... > Name: not available > Type: image/gif > Size: 2775 bytes > Desc: image001.gif > Url : > http://lists.digium.com/pipermail/asterisk-users/attachments/20070524/7908e840/attachment.gif > -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] WiFi SIP phones
Has anyone installed Linux on your ABP phones, and got all functionality (including GSM and WiFi)? Will these phones work in the US (which radio frequency modes)? On Thu, 2007-05-24 at 00:49 -0700, [EMAIL PROTECTED] wrote: > Date: Thu, 24 May 2007 00:10:23 -0500 > From: "Shanon Swafford" <[EMAIL PROTECTED]> > Subject: RE: [asterisk-users] WiFi SIP phones > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > > I work for ABP Technology and lurk on this list so I hope I'm not > breaking > any taboos... > > ABP is now carrying a dual GSM/Wifi phone. We tested 2 models, 1 had > Windows-CE on it. Some reason we only have the Non-CE version public > right > now. > > http://www.abptech.com/products/Pirelli/DPL10.html > http://www.abptech.com/products/Pirelli/DPL10.html> > http://var.abptech.com/s.nl/it.A/id.2041/.f?sc=2&category=31> > > VARs/Resellers/ITSPs/Consultants: > http://www.abptech.com/support/qa/index.php?target=become_reseller > http://www.abptech.com/support/qa/index.php?target=become_reseller > > > > End Users go here and we'll help you find a place to buy one: > http://www.abptech.com/aboutus/find_reseller.php > > Shanon > ABP Technology > -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Announcing - AstJax click2call Firefox greasemonkey script - click and dial phone numbers in any webpage
Is there any FireFox plugin that contains an entire (SIP or IAX) softphone, that can also be scripted in the page's HTML/Javascript? On Mon, 2007-05-21 at 06:20 -0700, [EMAIL PROTECTED] wrote: > Date: Mon, 21 May 2007 10:51:09 +0100 > From: "Richard Hamnett" <[EMAIL PROTECTED]> > Subject: [asterisk-users] Announcing - AstJax click2call Firefox > greasemonkey script - click and dial phone numbers in any > webpage > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > Hi there, > > Just to announce that I've improved upon a greasemonkey script which > allows > users to dial any number (in the given regex format) by turning it > into a > clickable hyperlink. > > The script uses greasemonkey's ajax callback to a simple php > controller > script, so that the click does not navigate away from the current > page. > > It requires an Asterisk Manager connection. > > See http://yaptele.com/asterisk-firefox-click-to-dial-ajax-script for > more > details. > > Kind Regards, > Richard Hamnett -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OpenWengo + Asterisk?
OpenWengo has just released WengoPhone v2.1.0: http://www.openwengo.org/index.php/openwengo/public/homePage/news?payload[newsId]=0 . Has anyone had success (or notable failures) using it as a client for Asterisk? Any advice on integrating it into dialplan, apps, config DBs, etc? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cpu usage for G.729 codec
(Note: resending with proper Subject) If I use Asterisk to initiate two call legs with a callfile, dialing the channel and setting the extension to an AGI that dials another channel, and both dial by SIP connection to a switch that allows only G.729, do I need a G.729 codec running on Asterisk? Do I need 2? And if I use the callfile to connect by SIP to a switch that allows only G.729, then use the extension AGI to play a file pre-encoded in G.729, do I need a codec? Where is the SW that encodes files in G.729? On Thu, 2007-05-17 at 08:38 -0700, [EMAIL PROTECTED] wrote: > Date: Thu, 17 May 2007 11:22:17 -0400 > From: "Race Vanderdecken" <[EMAIL PROTECTED]> > Subject: RE: [asterisk-users] cpu usage for G.729 codec > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > > G.729 is a compromise of bandwidth vs. CPU power. It takes more CPU > but > less bandwidth. > > It depends on what your want to do with the G.729. > > Pass through does not involve any transcoding, that I know of, so it > is > just an RTP packet movement, no different than the cost of other pass > through codecs. > > I did work on converting G.729 to G.711 to disk storage in real time > and > that took about 3% of a Xeon CPU for full duplex. > > Memory wise each convert call might have used 640KB in buffers and > trash, but not much. -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 34, Issue 82
If I use Asterisk to initiate two call legs with a callfile, dialing the channel and setting the extension to an AGI that dials another channel, and both dial by SIP connection to a switch that allows only G.729, do I need a G.729 codec running on Asterisk? Do I need 2? And if I use the callfile to connect by SIP to a switch that allows only G.729, then use the extension AGI to play a file pre-encoded in G.729, do I need a codec? Where is the SW that encodes files in G.729? On Thu, 2007-05-17 at 08:38 -0700, [EMAIL PROTECTED] wrote: > Date: Thu, 17 May 2007 11:22:17 -0400 > From: "Race Vanderdecken" <[EMAIL PROTECTED]> > Subject: RE: [asterisk-users] cpu usage for G.729 codec > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > > G.729 is a compromise of bandwidth vs. CPU power. It takes more CPU > but > less bandwidth. > > It depends on what your want to do with the G.729. > > Pass through does not involve any transcoding, that I know of, so it > is > just an RTP packet movement, no different than the cost of other pass > through codecs. > > I did work on converting G.729 to G.711 to disk storage in real time > and > that took about 3% of a Xeon CPU for full duplex. > > Memory wise each convert call might have used 640KB in buffers and > trash, but not much. -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail to Text Transcription(was: Re: [asterisk-dev] Voicemailto text translation)
(This subthread is more appropriate to -users than to -dev, so it is crossposted only to mark its transition. Please reply on the -user list only.) What are the cheapest prices for (humans) transcribing voicemail to text as a service? The absolute cheapest, regardless of (known) quality - the quality only has to compete with (cheaper) automated transcription, which is abysmal quality. On Wed, 2007-04-04 at 09:25 -0700, [EMAIL PROTECTED] wrote: > Date: Wed, 4 Apr 2007 09:25:02 -0700 > From: "Mike Taht" <[EMAIL PROTECTED]> > Subject: Re: [asterisk-dev] Voicemail to text translation > To: [EMAIL PROTECTED],"Asterisk Developers Mailing > List" > <[EMAIL PROTECTED]> > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > On 4/4/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> > wrote: > > > > Is anybody aware of a way to automate the translation or conversion > of > > voice mail files into text ? > > > Being that understanding random human speech at 8khz > > I had had a different idea. Merely have a voice mail option "press 4 > to > transcribe this" - which would take the vmail and ship it to a > transcription > service like "transcribr.com". There's a couple companies like that > that out > there do transcription - quite well, and cheaply. > > Sent via BlackBerry from T-Mobile > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-dev mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-dev > > > > > > -- > Mike Taht > PostCards From the Bleeding Edge > http://the-edge.blogspot.com -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source VoIP client (on a webpage)
On Fri, 2007-04-06 at 12:00 -0700, [EMAIL PROTECTED] wrote: > Date: Fri, 6 Apr 2007 16:13:29 +0100 > From: Tim Panton <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Open Source VoIP client (on a webpage) > To: Jason Wolfe <[EMAIL PROTECTED]>, Asterisk Users Mailing > List - Non-Commercial Discussion > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed > > > On 6 Apr 2007, at 00:59, Jason Wolfe wrote: > > > I need to decide on the best way to add a voip SIP or IAX client > to > > a website. I'm thinking that I'd like it to be inline, like an > > aplet, on the page. I've got some asterisk servers running to > > connect up to, so the real challenge is finding an easily > > integrated open source client. > > > > Any suggestions from those who know? > > Our SDK isn't open source, but it is an IAX applet - > javascript/DHTML > friendly and lightweight. Is that applet available unbundled from the rest of your software and service package? At a flat (ie not per-instance) price? > Tim Panton > > www.mexuar.net > www.westhawk.co.uk/ > -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail to Text Transcription(was: Re: [asterisk-dev] Voicemail to text translation)
(This subthread is more appropriate to -users than to -dev, so it is crossposted only to mark its transition. Please reply on the -user list only.) What are the cheapest prices for (humans) transcribing voicemail to text as a service? The absolute cheapest, regardless of (known) quality - the quality only has to compete with (cheaper) automated transcription, which is abysmal quality. On Wed, 2007-04-04 at 09:25 -0700, [EMAIL PROTECTED] wrote: > Date: Wed, 4 Apr 2007 09:25:02 -0700 > From: "Mike Taht" <[EMAIL PROTECTED]> > Subject: Re: [asterisk-dev] Voicemail to text translation > To: [EMAIL PROTECTED],"Asterisk Developers Mailing > List" > > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > On 4/4/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> > wrote: > > > > Is anybody aware of a way to automate the translation or conversion > of > > voice mail files into text ? > > > Being that understanding random human speech at 8khz > > I had had a different idea. Merely have a voice mail option "press 4 > to > transcribe this" - which would take the vmail and ship it to a > transcription > service like "transcribr.com". There's a couple companies like that > that out > there do transcription - quite well, and cheaply. > > Sent via BlackBerry from T-Mobile > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-dev mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-dev > > > > > > -- > Mike Taht > PostCards From the Bleeding Edge > http://the-edge.blogspot.com -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: On Topic: Cheapest Asterisk USB Key?
On Mon, 2007-04-02 at 16:30 -0700, [EMAIL PROTECTED] wrote: > Date: Mon, 02 Apr 2007 20:26:09 +0100 > From: Thomas Kenyon <[EMAIL PROTECTED]> > Subject: [asterisk-users] Re: On Topic: Cheapest Asterisk USB Key? > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1 > > Salvatore Giudice wrote: > > > You should be aware that flash memory is generally not the best > medium to > > store data when you have a high number of read/writes. Flash memory > will > > fail much more quickly under these conditions. > > Does this mean that devices such as the samsung Flash SSD (part # > MCAQE32G5APP-0XA00) and the Supertalent Flashdrives are less reliable > than the HD equivalents. (since reliability is supposed to be their > biggest selling points)? What it means is that Flash memory cells wear out after a large number of read/write cycles, but not nearly as large as hard drives: http://en.wikipedia.org/wiki/Flash_rom#Limitations . So using Flash in place of RAM, even when high speed isn't important, can wear out the Flash - it will probably wear out even before HDs, which live less long than does RAM. Until the Flash wears out, it is extremely reliable, and techniques for ensuring it doesn't destroy data as it wears out are built into the Flash HW (though it will eventually wear out take data with it). But I'm not talking about using the Flash as RAM, just using it for a low-load persistent store like a HD, where a HD would be overkill in every way. -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: On Topic: Cheapest Asterisk USB Key?
I need a USB microprocessor *device* on which the Linux and Asterisk will run (even if very slowly), not just a storage drive from which to run it on the PC. MonteVista is a good distro, though there are other "minimal" embedded distros, of which I've already got one selected. The CDR usage of a single user's PC is just fine in performance and total lifetime read/writes (usually upwards of 100K) for the CDR data that needs to persist, as opposed to the device's RAM for executing the Asterisk. I'm looking for a device under $100 or $50 in OEM quantity, which is where just microdrives start. I want to run Asterisk itself, even if stripped down, for easy sync and single platform maintenance across all the Asterisk instances I've got, as well as guaranteed compatibility between data/network formats/protocols. On Sun, 2007-04-01 at 13:08 -0400, Salvatore Giudice wrote: > Try installing Monte Vista http://www.mvista.com/ on the usb stick. It will > be a lot cleaner than taking a standard server distribution of linux and > stripping out all the unwanted kernel modules. > > Monte Vista is an embedded linux that should be able to boot your server off > a 128mb usb stick with Asterisk installed. You should probably strip > asterisk down to the bare essentials for your project as well. > > You should be aware that flash memory is generally not the best medium to > store data when you have a high number of read/writes. Flash memory will > fail much more quickly under these conditions. You might want to conside > using a usb microdrive instead of a flash stick. Pick a microdrive that > generates as little heat as possible. > > BTW, what exactly is the motivation for running linux off of a usb stick? If > you would like cdr's, you could likely do so with ngrep and a perl script. > > Good luck, SG > > -- > Salvatore Giudice > [EMAIL PROTECTED] > > VoIP Security Training, LLC > http://VoIPSecurityTraining.com > > 848 N. Rainbow Blvd. #1676 > Las Vegas, NV 89107 > Phone: (702) 979-2906 > Fax: (212) 279-2906 > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Matthew > Rubenstein > Sent: Sunday, April 01, 2007 9:08 AM > To: Asterisk-Users > Subject: On Topic: Cheapest Asterisk USB Key? (was: Re: [asterisk-users] Off > Topic: Open Source USB Softphone) > > Here's a flipside of this subject: what is the absolute cheapest > Linux > device that can be connected to a PC's USB port? That has just enough > power for a minimal Asterisk server running on it. The Asterisk just > maintains a CDR database on its Flash memory, which it periodically > submits over the PC's network connection with an HTTP hit on a remote > full-service Asterisk server? No call handling, DSP or anything really > number crunching, no telephony terminal or other services. The > lowest-performance device that plugs into the USB, with its own Linux > instance. In OEM quantity, under $50? Under $100? > > > On Sun, 2007-04-01 at 02:51 -0700, > [EMAIL PROTECTED] wrote: > > Date: Sat, 31 Mar 2007 16:02:06 -0500 > > From: "Mike Lynchfield" <[EMAIL PROTECTED]> > > Subject: Re: [asterisk-users] Off Topic: Open Source USB Softphone > > To: [EMAIL PROTECTED], "Asterisk Users Mailing List - > > Non-Commercial > > Discussion" > > Message-ID: > > <[EMAIL PROTECTED]> > > Content-Type: text/plain; charset="iso-8859-1" > > > > sip would be the required one as iax..well.. > > > > also openwengo wont work.. to much overhead .. broswrer needed.. ie > > component + flash + css+js etc.. not viable.. > > > > so im also asking anyone have one ? since ihave a supply of around > > 2000 of > > the vonage usb stick OEM.. > > > > On 3/30/07, Michael Van Donselaar <[EMAIL PROTECTED]> > > wrote: > > > > > > Which USB Phone? I have written custom versions of iaxcomm for > > various > > > people, > > > and have a version that works with the Yealink phone. > > > > > > On Thu, 29 Mar 2007 11:33:07 -0300, "Luis Claudio Santos" < > > > [EMAIL PROTECTED]> > > > wrote: > > > > > > >I need a softphone - for usb phone devices - that I can alter > > (insert > > > logo, > > > >menu, etc). > > > > > > > >Does somebody know such one? > > > > > > > >[]s > > > > > > ___ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > -- > > Mike > > Sales Manager > > http://www.voicemeup.com > > Making it happen > > 1.877.807.VOIP (8647) > > 1.514.312.7030 -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
On Topic: Cheapest Asterisk USB Key? (was: Re: [asterisk-users] Off Topic: Open Source USB Softphone)
Here's a flipside of this subject: what is the absolute cheapest Linux device that can be connected to a PC's USB port? That has just enough power for a minimal Asterisk server running on it. The Asterisk just maintains a CDR database on its Flash memory, which it periodically submits over the PC's network connection with an HTTP hit on a remote full-service Asterisk server? No call handling, DSP or anything really number crunching, no telephony terminal or other services. The lowest-performance device that plugs into the USB, with its own Linux instance. In OEM quantity, under $50? Under $100? On Sun, 2007-04-01 at 02:51 -0700, [EMAIL PROTECTED] wrote: > Date: Sat, 31 Mar 2007 16:02:06 -0500 > From: "Mike Lynchfield" <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Off Topic: Open Source USB Softphone > To: [EMAIL PROTECTED], "Asterisk Users Mailing List - > Non-Commercial > Discussion" > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > sip would be the required one as iax..well.. > > also openwengo wont work.. to much overhead .. broswrer needed.. ie > component + flash + css+js etc.. not viable.. > > so im also asking anyone have one ? since ihave a supply of around > 2000 of > the vonage usb stick OEM.. > > On 3/30/07, Michael Van Donselaar <[EMAIL PROTECTED]> > wrote: > > > > Which USB Phone? I have written custom versions of iaxcomm for > various > > people, > > and have a version that works with the Yealink phone. > > > > On Thu, 29 Mar 2007 11:33:07 -0300, "Luis Claudio Santos" < > > [EMAIL PROTECTED]> > > wrote: > > > > >I need a softphone - for usb phone devices - that I can alter > (insert > > logo, > > >menu, etc). > > > > > >Does somebody know such one? > > > > > >[]s > > > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > -- > Mike > Sales Manager > http://www.voicemeup.com > Making it happen > 1.877.807.VOIP (8647) > 1.514.312.7030 -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Viruses?
The Skype network is circulating a virus that has appeared there before: http://www.informationweek.com/news/showArticle.jhtml?articleID=198500135 . The virus sends a URL to other Skype users in the infected user's contacts, which the target Skype displays as clickable. Clicking downloads the virus. Asterisk supports features like these, in combination with certain clients (which aren't themselves Asterisk), including IM and URL redirection. Any reports of this kind of attack on Asterisk itself, or using Asterisk to support those potentially vulnerable clients? Any analysis of Asterisk's vulnerability to these? Any mitigations? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme - is this statement from the Wiki still true?
I don't know whether the Asterisk default codec is still ulaw or not, though I believe it is. But it doesn't matter. Any connection in the same call between multiple legs which each use a different codec, whether GSM or otherwise, will of course require Asterisk to consume CPU in transcoding between the two different encodings. Each leg gets a codec which transcodes to Asterisk's native encoding (which I believe is ulaw), in which encoding mixing and other signal processing is performed, before reencoding back into the encoding each leg uses. On Thu, 2007-02-15 at 15:31 -0700, [EMAIL PROTECTED] wrote: > Date: Fri, 16 Feb 2007 08:42:49 +1100 > From: "Eric Bishop" <[EMAIL PROTECTED]> > Subject: [asterisk-users] Meetme - is this statement from the Wiki > still true? > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > "The conference bridge runs Ulaw codec by default. If you let people > connect > with GSM or other codecs, Asterisk will use CPU power to convert audio > between codecs" ... What about alaw channels is there any transcoding > work > being done there? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi-calendar Overlay Layers?
Sorry, I sent that message to the wrong list. Tho if you know the answer, please don't let that stop you from emailing it to me :). On Thu, 2007-02-15 at 08:21 -0700, [EMAIL PROTECTED] wrote: > Date: Thu, 15 Feb 2007 08:54:43 -0500 > From: Matthew Rubenstein <[EMAIL PROTECTED]> > Subject: [asterisk-users] Multi-calendar Overlay Layers? > To: Asterisk-Users > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain > > Is there any calendar client that can point at OX for calendar > data, > which client can display multiple calendars simultaneously as > *overlapping layers* in the GUI? With UI to de/select calendars from > view, one by one. That is, a single grid of days displayed, with the > events in each day displayed in the same day's view list, as if the > layers were all events in a single calendar. > > And is there a way to get the OX Web interface to do this? Or > a place > in the source code that can be recoded to do it? Thanks. -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multi-calendar Overlay Layers?
Is there any calendar client that can point at OX for calendar data, which client can display multiple calendars simultaneously as *overlapping layers* in the GUI? With UI to de/select calendars from view, one by one. That is, a single grid of days displayed, with the events in each day displayed in the same day's view list, as if the layers were all events in a single calendar. And is there a way to get the OX Web interface to do this? Or a place in the source code that can be recoded to do it? Thanks. -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and multiple cpus/cores
Yes, that is perfectly clear - thanks for the data. The problem with discussing load capacity of hosts running codecs is just how many codecs are running at a time, how many code/decode instances each "call" comprises, without knowing how many codecs are running per "call". I'm disappointed that Digium has not published exhaustive benchmarks on capacity planning on different HW configs for different running setups. An major unanswered question in planning Asterisk deployments, especially scalable biz apps, is "how many hosts with what specs will I need for X, Y, Z scales and usage combos?" This question is asked several times in different ways on the maillists every month. And of course the answers aren't in terms that can be collated into a consistent planning guide. Since Asterisk is free, and minutes are cheap, the HW, though relatively (to proprietary) cheap, is still a major cost fraction. And of course running out of capacity by surprise is a crippling blow. On Sat, 2007-02-10 at 15:57 -0500, Andres wrote: > Matthew Rubenstein wrote: > > > Are there 45 G.729 instances for the 45 ZAP legs in addition to 45 > >G.729 instances for the 45 SIP legs? Or do the ZAP legs not get a codec > >(HW instead)? > > > > > > > Its a 4 Port T1 with 92 ZAP channels. So we are talking about 90 SIP > Channels being fed into 90 ZAP channels (which means 180 people are > talking, 90 on SIP Phones and 90 on PSTN lines). We are therefore > transcoding 90 G.729 calls into 90 G.711 Calls. It eats up 90 G.729 > licenses. I hope that clears things up. > > Digium has also reported 80 G729 calls on their own dual cpu 2.8Ghz > Xeon boxes: http://www.digium.com/en/products/voice/g729codec.php > > >On Sat, 2007-02-10 at 12:06 -0500, Andres wrote: > > > > > >>Hi Matthew, > >> > >>Yes, those are really 90 SIP-ZAP calls. Which means the 4 port T1 is > >>pretty much full of calls. All SIP endpoints are forced to G729. And > >>as for your 125% question I really don't know why. This is just what I > >>can see from our MRTG graphs. We graph all CPU usage and SIP/ZAP > >>calls. All our servers are running Asterisk 1.2.9.1. > >> > >>Andres. > >> > >> > >>Matthew Rubenstein wrote: > >> > >> > >> > >>> Are those "90 calls" really 90 instances of the G.729 codec (+ other > >>>processing), 90 "legs" (people at phones) for 45 2-party calls? > >>> > >>> Also, how do you get 125% more CPU bandwidth by adding another CPU, > >>>which usually gets less than 100% more power after its overhead to > >>>function in the system? > >>> > >>> > >>>On Sat, 2007-02-10 at 04:46 -0700, > >>>[EMAIL PROTECTED] wrote: > >>> > >>> > >>> > >>> > >>>>Date: Fri, 09 Feb 2007 21:57:23 -0500 > >>>>From: Andres <[EMAIL PROTECTED]> > >>>>Subject: Re: [asterisk-users] asterisk and multiple cpus/cores > >>>>To: Asterisk Users Mailing List - Non-Commercial Discussion > >>>> > >>>>Message-ID: <[EMAIL PROTECTED]> > >>>>Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >>>> > >>>>Erick Perez wrote: > >>>> > >>>> > >>>> > >>>> > >>>> > >>>>>I have found a site that list the following (no date in the post, so > >>>>>it may be old): > >>>>>"since all transcoding and calls still go through one core in > >>>>> > >>>>> > >>>>> > >>>>> > >>>>asterisk, > >>>> > >>>> > >>>> > >>>> > >>>>>it doesn't make sense to buy a multi-core or hyperthreaded system > >>>>> > >>>>> > >>>>> > >>>>> > >>>>that > >>>> > >>>> > >>>> > >>>> > >>>>>will only slow you down" > >>>>> > >>>>>Does that still applies in asterisk 1.2.14/1.4.x ? > >>>>>Or do we have to tweak source code to balance loads > >>>>> > >>>>> > >>>>> > >>>>> > >>>>(transcoding,etc) > >>>> > >>>> > >>>> > >>>> > >>>>>between cores? > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>I can tell you that statement is bogus. We run a number of dual cpu > >>>>and > >>>>single cpu systems on our network. The dual ones (Xeon 3.6Ghz) can > >>>>easily handle 90 G729 calls at 50% CPU Usage. The single ones will > >>>>be > >>>>at 50% with only 40 calls. > >>>> > >>>>Andres > >>>> > >>>> > >>>> > >>>> > -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and multiple cpus/cores
Are there 45 G.729 instances for the 45 ZAP legs in addition to 45 G.729 instances for the 45 SIP legs? Or do the ZAP legs not get a codec (HW instead)? On Sat, 2007-02-10 at 12:06 -0500, Andres wrote: > Hi Matthew, > > Yes, those are really 90 SIP-ZAP calls. Which means the 4 port T1 is > pretty much full of calls. All SIP endpoints are forced to G729. And > as for your 125% question I really don't know why. This is just what I > can see from our MRTG graphs. We graph all CPU usage and SIP/ZAP > calls. All our servers are running Asterisk 1.2.9.1. > > Andres. > > > Matthew Rubenstein wrote: > > > Are those "90 calls" really 90 instances of the G.729 codec (+ other > >processing), 90 "legs" (people at phones) for 45 2-party calls? > > > > Also, how do you get 125% more CPU bandwidth by adding another CPU, > >which usually gets less than 100% more power after its overhead to > >function in the system? > > > > > >On Sat, 2007-02-10 at 04:46 -0700, > >[EMAIL PROTECTED] wrote: > > > > > >>Date: Fri, 09 Feb 2007 21:57:23 -0500 > >>From: Andres <[EMAIL PROTECTED]> > >>Subject: Re: [asterisk-users] asterisk and multiple cpus/cores > >>To: Asterisk Users Mailing List - Non-Commercial Discussion > >> > >>Message-ID: <[EMAIL PROTECTED]> > >>Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >> > >>Erick Perez wrote: > >> > >> > >> > >>>I have found a site that list the following (no date in the post, so > >>>it may be old): > >>>"since all transcoding and calls still go through one core in > >>> > >>> > >>asterisk, > >> > >> > >>>it doesn't make sense to buy a multi-core or hyperthreaded system > >>> > >>> > >>that > >> > >> > >>>will only slow you down" > >>> > >>>Does that still applies in asterisk 1.2.14/1.4.x ? > >>>Or do we have to tweak source code to balance loads > >>> > >>> > >>(transcoding,etc) > >> > >> > >>>between cores? > >>> > >>> > >>> > >>I can tell you that statement is bogus. We run a number of dual cpu > >>and > >>single cpu systems on our network. The dual ones (Xeon 3.6Ghz) can > >>easily handle 90 G729 calls at 50% CPU Usage. The single ones will > >>be > >>at 50% with only 40 calls. > >> > >>Andres > >> > >> > -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dependencies on DB?
What are the specific dependencies that Asterisk has on databases? Some hi-perf data is stored in BDB, CDRs are in a relational DB like MySQL. Is there a list of specific dependencies by specific modules on specific tables? A complete list, so switching from the default DB can drop the old DB from the install. -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Cmd to ID Mobile from Phone#?
Is there an Asterisk command, app, AGI (or other) that can be called with a phone# (or list) that will lookup somewhere definitive and report whether the phone# is registered to a mobile phone or not? How about other data, like its home city/district etc? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which Java FastAGI implementation has the most "market share"?
On Mon, 2007-02-05 at 04:46 -0700, [EMAIL PROTECTED] wrote: > > Date: Sun, 04 Feb 2007 23:35:46 -0500 > From: Steve Prior <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Which Java FastAGI implementation has > the most"market share"? > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Kate Kretz wrote: > > Steve, keep me in touch please ? > > We are also looking for moving all our activities to java platform. > > > > Let me know if You'll find/test something like asterisk2billing > written > > in java ? > > I haven't received any feedback at all on the relative use of the java > options, but I'm pretty happy with the way a little project turned > out > in asterisk-java. > > My project was to see how well asterisk-java would work in > combination > with Lumenvox to create a speech enabled AGI, so just for kicks I've > ported their Pizza ordering demo to Java using it. In the process > I've > been working with Lumenvox to fix the couple of problems which turned > up > as a result of this experiment, and use an asterisk-java code change > which is available in their latest svn. > > Sometime soon my code will be made available most likely through the > Lumenvox site so others can use it as a starting point. > > Overall I'll say that I really like using Java to control such a dial > plan. In this particular case the output is a simple pizza order > which > I've modeled as a plain old Java object (POJO), so once the dial plan > has built up the object it can simply be passed to whatever back end > (possibly J2EE) code which processes the transaction without regard > for > the user interface that created it. Sounds very maintainable to me. > I > did the development/test right in the Eclipse IDE and could use the > debugger when necessary - I've got to believe that's better than > trying > to trace through a regular dial plan. > > I also really like the fact that aside from sound files and just a > couple of lines of dial plan code to call the Java, all the actual > Java > code is running in a different server box so I'm keeping the load > down > on my Asterisk box and have flexibility in where I deploy things. The real advantage in choosing an AGI (or CGI or ...) platform/language is *reusing* the existing code that already runs on that platform, with minimal porting to the platform in that language. How much does a Java application, net/bean, or modern (1.4-6.x) class have to be revised to make it work with asterisk-java as FastAGI instead of, say, AGI, CGI, commandline, browser JVM, or other execution environment/UI? > Steve > > > > > Cheers, > > Kate > -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which Java FastAGI implementation has the most "market share"?
On Mon, 2007-02-05 at 23:05 -0500, Steve Prior wrote: > > Matthew Rubenstein wrote: > > > > The real advantage in choosing an AGI (or CGI or ...) platform/language > > is *reusing* the existing code that already runs on that platform, with > > Well of course you should pick whatever AGI implementation matches the > rest of your environment best. > > > minimal porting to the platform in that language. How much does a Java > > application, net/bean, or modern (1.4-6.x) class have to be revised to > > make it work with asterisk-java as FastAGI instead of, say, AGI, CGI, > > commandline, browser JVM, or other execution environment/UI? > > I'm not totally sure you're asking the right question here. > Asterisk-java in combination with Asterisk and in my case Lumenvox is > just a user interface for whatever application I am developing. In my > case it's not even the only user interface I've created for my system > (which happens to be in Home Automation which uses CORBA to connect the > pieces together) - I've also got a web interface as well as other > standalone front ends and even the light switches can be considered part > of the UI (and therefore non reusable). Asterisk-java provides you with > an ordinary JRE environment where you might not be in direct control of > main() (though you can be if you really want to), but that's similar to > the other server environments you mentioned (browser JVM is a different > animal). > > So the real question isn't so much how a class needs to be revised for > asterisk-java, it is does your back end system provide a robust API such > that you can be dropped naked in the middle of a JRE woods and without > anything more than some additions to the CLASSPATH be able to interact > with your back end system. So you're saying that if you're using Sun's JRE 1.6.0 in Tomcat full of existing classes connected into apps, that pointing Asterisk's FastAGI at it just requires asterisk-java on the Asterisk server and adding a very simple FastAGI wrapper class to the Tomcat server to interface Asterisk's runtime state to the existing apps. And that a FastAGI wrapper class will also work on just Apache running a java commandline CGI, etc. > Steve > -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Callfiles to Meetme Fails (was: RE: [asterisk-users] Using Local Channels with Originate)
I have Meetme conferencing compiled for Debian as per
http://powerontech.com/freepbx-on-debian.htm . I drop a callfile in the
outgoing dir, and it intitiates a call to a local extension as a
channel, but the call seems to block on the Meetme() command. That
extension completes the outgoing Dial(SIP) command to my phone,
announcing that leg is the only member of the conference, and just
waits. If I hang up, the exit status is reported by Asterisk (in logfile
and CLI), then Asterisk jumps to the callfile's extension, which
completes the outgoing Dial(SIP) to the other phone, which then gets the
announcement that it is the only member of that conference. Why does it
block, instead of proceeding to the second leg and conferencing it in?
meetme.conf :-
[rooms]
conf => 1234
EOF
from extensions.conf :-
[ext-out]
exten => callFrom,1,Noop(Calling SIP/[EMAIL PROTECTED])
exten => callFrom,n,Dial(SIP/[EMAIL PROTECTED],45,M(conf-from^
${callTo})g)
exten => callFrom,n,Noop(Done dialing from)
exten => callTo,1,Noop(Calling SIP/[EMAIL PROTECTED])
exten => callTo,n,Dial(SIP/[EMAIL PROTECTED],45,M(conf-to^999)g)
exten => callTo,n,Noop(Done dialing to)
[macro-conf-from]
; ARG1: callTo
exten => s,1,Noop(in macro-conf-from)
exten => s,n,Noop(before MeetMe: ${ARG1})
exten => s,n,MeetMe(1234)
exten => s,n,Noop(after MeetMe: ${ARG1})
EOF
out.call :-
Channel: Local/[EMAIL PROTECTED]/n
Callerid: 12126661234
Context: ext-out
Extension: callTo
Priority: 1
Set: callFrom=12126661234
Set: callTo=1212777
Set: callerID=12126661234
Set: sipCarrier=sipcarrier
EOF
On Mon, 2007-02-05 at 15:52 -0700,
[EMAIL PROTECTED] wrote:
> Date: Mon, 05 Feb 2007 17:37:31 -0500
> From: David Boyd <[EMAIL PROTECTED]>
> Subject: RE: [asterisk-users] Using Local Channels with Originate
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset=utf-8
>
> On Mon, 2007-02-05 at 13:21 -0800, Michael Collins wrote:
> > I havent quite gotten this working yet but I am going to update the
> > thread with what I have learned. Maybe this will help the next guy
> who
> > tries to figure this out
> >
> >
> >
> > The trick to using the DIALSTATUS seems to be to put it in the
> handler
> > for the h (hang-up extension).
> >
> >
> >
> > [outdialer]
> >
> > exten => 100, 1, Dial(${numberToDial})
> >
> > exten => h, 1, Goto(s-${DIALSTATUS},1)
> >
> >
> >
> > exten => s-ANSWER,1,NoOp("Answered")
> >
> > exten => s-BUSY,1,NoOp("Busy")
> >
> > exten => s-NOANSWER,1,NoOp("Not answered")
> >
> > exten => s-CANCEL,1,NoOp("Cancelled")
> >
> > exten => s-CONGESTION,1,NoOp("Fast busy")
> >
> > exten => s-CHANUNAVAIL,1,NoOp("Channel unavailable")
> >
> >
> >
> > [dialerplan]
> >
> > exten => s,1,Background(demo-congrats)
> >
> > exten => s,n,WaitExten
> >
> > so on ...
> >
> >
> >
> > Here are the manager commands I am using:
> >
> >
> >
> > Action: login
> >
> > Username: test
> >
> > Secret: nottelling
> >
> >
> >
> > Action: originate
> >
> > Channel: Local/[EMAIL PROTECTED]/n
> >
> > Context: dialerplan
> >
> > Extension: s
> >
> > Priority: 1
> >
> > Variable: numberToDial=ZAP/4/1234567890
> >
> >
> >
> > Action: logoff
> >
> >
> >
> > I am always getting ANSWERED for ${DIALSTAUS} so something is not
> > quite right. Hopefully I am getting closer.
> >
> >
> >
> >
> >
> > Brian,
> >
> >
> >
> > What kind of Zap hardware/telco lines are you using? I am using PRI
> > and I am able to get a dial status in the hangup extension. The
> > problem I run into is that I get NO ANSWER as the hangup cause even
> > for invalid phone numbers I also get cluttered CDRs. In the
> > meantime Im working on a solution that I hope will give the
Re: [asterisk-users] Question on G.729
On Mon, 2007-02-05 at 12:00 -0700, [EMAIL PROTECTED] wrote: > Date: Mon, 5 Feb 2007 11:36:28 -0500 > From: Andy Davidson <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Question on G.729 (was: H.264 *Not > Patented*) > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed > > > On 1 Feb 2007, at 14:14, Lacy Moore - Aspendora wrote: > > > On 2/1/07, Andy Davidson <[EMAIL PROTECTED]> wrote: > > > What I would expect to happen, is that Asterisk would transcode > > > between the ulaw/alaw party, and me, wanting to listen via > g729. Is > > > this what *should* happen ? Worth noting that my provider does > not > > > support G.729. Is what is happening a bug ? Any patches I can > try > > > to see if they work ? Or is it my config which is broken ? > > How many g729 licenses do you have? > > Just one - my interpretation was that one license bought one > inbound, > and one outbound transcoder, so my scenario would work with this > (phone and * talk g.729, then * turns g.729 into ulaw for my > upstream..) > > Do I need to buy more licenses ? The consensus on this list is that Digium G.729 licenses apply to *each running instance* that is either encoding or decoding. Which means each *leg* of a call, if it is being transcoded, whether that is a single caller in a multi-caller (eg, 2 people or more) or even an app. So if both people in a call are sending G.729 encoded data, and your app decodes the *mixed* G.729 into ulaw (or slinear or any other decoded format it outputs) requiring a single instance of the decoder, then you need a single license. Multiple simultaneous calls working exactly like that each need a single license, #licenses = #calls. But if your app decodes both G.729 legs into ulaw (or other working format) data that is then mixed or otherwise processed, then the two simul codecs for the two legs need two licenses. > -a > -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FreePBX/Debian Aborts Call While Connecting
You're looking at only the logfiles, which don't reflect the problem at
the other side, the switch which sees the incoming request abort before
it can complete the connection, and before the 45s timeout. What you're
missing is my reports of that difference on either side of the network,
which I have mentioned in every message to this list, including the one
you counted.
In any event, the problem is some kind of protocol handling bug, either
in the SIP server or the (SVN) version of Asterisk I'm using. I pointed
at a different (newer) SIP server at my same carrier, and have no
problem connecting. Though I was connecting OK to the old SIP server
with my old Asterisk version (1.2.12) before the "upgrade". I expect
that both the old SIP server and the SVN Asterisk version have bugs
which finally combine to abort improperly, and without proper failure
reporting by Asterisk.
Thanks anyway for trying to help.
On Thu, 2007-02-01 at 22:59 -0500, Asterisk wrote:
> On Thu, 2007-02-01 at 08:47 -0500, Matthew Rubenstein wrote:
> > The point is that the SIP carrier side gets the abort *before the SIP
> > carrier can complete the connection*. That doesn't take 45s. It takes
> > something like a few seconds. What is causing my (Asterisk) side to
> > abort right after completing registration?
> >
> >
> > On Thu, 2007-02-01 at 02:28 -0500, Asterisk wrote:
> > > Yeah, your waittime parameter in your call file is set to 45 seconds.
> > >
> > > db
> > >
> > > On Wed, 2007-01-31 at 21:52 -0500, Matthew Rubenstein wrote:
> > > > I used the "FreePBX on Debian" HowTo at
> > > > http://powerontech.com/freepbx-on-debian.htm to install. I use callfiles
> > > > to initiate calls to my SIP carrier. They get my registration, but they
> > > > see that my call is interrupted before they can complete the connection.
> > > > My Asterisk log shows that the call times out after the time (45s)
> > > > specified in my dialplan Dial() command. What is wrong?
> > > >
> > > > [from /var/log/asterisk/full]:
> > [...]
>
> Alright, take a look the **Lines:
>
>
>
> **Line 1:
> Your dial sequence clearly shows the 45sec timeout value being applied
> as the second value in the dial plan "SIP/[EMAIL PROTECTED]|45| <<--
>
> Jan 30 23:41:30 VERBOSE[17269] logger.c: -- Executing
> Dial("Local/[EMAIL PROTECTED],2", "SIP/[EMAIL PROTECTED]|45|
> M(say-call-2-digits^17182335097)g") in new stack
>
>
> **Line 2:
> The timer has expired 45000ms is the same 45 second timer that was set
> for timeout
>
> Jan 30 23:42:15 VERBOSE[17269] logger.c: -- Nobody picked up in
> 45000 ms
>
> Line 3:
> The call is dropped towards the carrier.
>
>
> Maybe I am missing something here but it seems you are using a macro
> with some global variable set for a 45 second wait time for outbound
> calls.
>
>
> Thanks,
> Dave
>
--
(C) Matthew Rubenstein
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] FreePBX/Debian Aborts Call While Connecting
The point is that the SIP carrier side gets the abort *before the SIP carrier can complete the connection*. That doesn't take 45s. It takes something like a few seconds. What is causing my (Asterisk) side to abort right after completing registration? On Thu, 2007-02-01 at 02:28 -0500, Asterisk wrote: > Yeah, your waittime parameter in your call file is set to 45 seconds. > > db > > On Wed, 2007-01-31 at 21:52 -0500, Matthew Rubenstein wrote: > > I used the "FreePBX on Debian" HowTo at > > http://powerontech.com/freepbx-on-debian.htm to install. I use callfiles > > to initiate calls to my SIP carrier. They get my registration, but they > > see that my call is interrupted before they can complete the connection. > > My Asterisk log shows that the call times out after the time (45s) > > specified in my dialplan Dial() command. What is wrong? > > > > [from /var/log/asterisk/full]: [...] -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FreePBX/Debian Aborts Call While Connecting
an 30 23:47:44 DEBUG[6245] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 615: Match Found Jan 30 23:47:44 DEBUG[6245] chan_sip.c: Registration successful Jan 30 23:47:44 DEBUG[6245] chan_sip.c: Cancelling timeout 17864 Jan 30 23:48:16 DEBUG[6245] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Jan 30 23:48:36 DEBUG[6245] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Jan 30 23:48:44 DEBUG[6268] manager.c: Manager received command 'Command' Jan 30 23:48:44 DEBUG[6268] manager.c: Manager received command 'Command' Jan 30 23:49:29 DEBUG[6245] chan_sip.c: Scheduled a registration timeout for 66.153.22.16 id #17872 Jan 30 23:49:29 DEBUG[6245] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 616: Found Jan 30 23:49:29 DEBUG[6245] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 616: Match Found Jan 30 23:49:29 DEBUG[6245] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 617: Found Jan 30 23:49:29 DEBUG[6245] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 617: Match Found Jan 30 23:49:29 DEBUG[6245] chan_sip.c: Registration successful Jan 30 23:49:29 DEBUG[6245] chan_sip.c: Cancelling timeout 17872 Jan 30 23:49:36 DEBUG[6245] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Jan 30 23:50:01 DEBUG[6245] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Jan 30 23:50:36 DEBUG[6245] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Jan 30 23:50:44 DEBUG[6268] manager.c: Manager received command 'Command' Jan 30 23:50:44 DEBUG[6268] manager.c: Manager received command 'Command' -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] H.264 *Not Patented*
The H.264 codec patent by Qualcomm has been ruled invalid by a San Diego Federal jury: http://www.eetimes.com/news/semi/showArticle.jhtml?articleID=197001066 . That means that H.264 codecs can now be written, distributed and revised freely under any license their authors choose, including GPL, public domain, or any other, and $free now that royalties are no longer required. How does H.264 compare with GSM and G.729 in CPU demand (MIPS:Kbps) and audio quality at low bitrates? GSM is $free, but G.729 is higher quality (tho patented with at least $10 per running codec instance royalties). Will H.264 become the favorite high-quality Asterisk codec, or will it perhaps force G.729 to become free, or negligibly cheaper? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CPU Bandwidth Consumption
Is the Asterisk processing and mixing of SIP channels into a single call (simple/minimum, no transcoding etc) calculated in integer or floating point instructions? How much CPU bandwidth is used per call leg, in either MIPS or MFLOPS? How about the G.729 codec, or other codecs: MIPS/MFLOPS? Any ideas how efficient is the Asterisk/x86 code compared to the maximum in the algorithm, that either SW optimization or porting to a more efficient processor (or both) could produce? -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] postgres and asterisk
Is that procedure the way to completely switch Asterisk from dependency on MySQL to dependency on Postgres instead? How about with Asterisk 1.4? And anyone have any idea whether FreePBX can be switched from MySQL to Postgres, too? On Tue, 2007-01-09 at 16:01 -0700, [EMAIL PROTECTED] wrote: > Date: Tue, 9 Jan 2007 16:54:24 -0400 > From: "Humberto Figuera" <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] postgres and asterisk > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi O.Youssef, > > if you asterisk version is 1.2.X > > edit apps/Makefile > > and discomment the line that contain 'app_sql_postgres.so': > > # > # Obsolete things... > # > APPS+=app_sql_postgres.so > #APPS+=app_sql_odbc.so > > save > > if you use debian: > > aptitude install libpq-dev > > and compile again > > I hope this be helpfull ;p > > -- > Humberto Figuera - Using Linux 2.6.18 > Usuario GNU/Linux 369709 > Caracas - Venezuela > GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA 37AD 3364 01D1 74CA > 0603 > > -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some queries on g729 license.
Did you find any operations trouble installing/using the Digium codec with Asterisk? I'd be surprised if Digium's were hard to use with Asterisk, considering they wrote and support both. Also can their codec be used to pre-encode data to files from a Linux command/line? Or just the Asterisk CLI mentioned earlier in this thread? On Tue, 2007-01-09 at 00:31 +0200, Zoa wrote: > I did some tests a long time ago and the speed was roughly the same. ( I > think digium's was slightly faster). > I think the IPP version also doesn't work on AMD out of the box. > > It's just 10$ a channel, that's not even worth the hassle of trying > something else. > > Joachim > > Al Bochter wrote: > > Matthew > > > > I agree. I only know what I have told by others so I do need this input > > > > I have been told that Digum G729 is a big pain the the butt to get > > working with Asterisk > > and it is very hard on the CPU > > > > Keep in mind I have never used any Ver. of G 729 > > > > So tell me what you think. > > > > Best regards, > > > > Al Bochter > > Bochter Services > > http://www.BochterServices.com/?t=Email > > > > > > > > Matthew Rubenstein wrote: > > > >> All of which hassle and expense can be avoided by buying a > >> license for > >> Digium's codec, which is tested to work well with Asterisk (and might > >> come with some support). And is pretty cheap per simul "call". > >> > >> I wonder whether that "per call" means "per codec instance", which > >> could be multiple licenses on a single conference call, where multiple > >> (even if not all) parties are getting de/encoded simultaneously. And > >> whether there are other tools for editing (/mixing/transforming) g729 > >> data, in realtime (streams) or not (files), and whether they require a > >> license. Ideally sox or equivalent would work on g729, maybe with a > >> codec plugin. > >> > >> > >> On Mon, 2007-01-08 at 13:23 -0500, Paul wrote: > >> > >> > >>> First point to tackle in any case involving patent, copyright or > >>> trademark infringement is whether or not the infringing party would > >>> have > >>> been qualified to buy any usage rights at all. In a case where you > >>> license the Intel source(read the terms, it's not really that "free"), > >>> you would be applying for a license under some plan that includes > >>> certain minimum payments. Even if you wrote new source from scratch you > >>> would be in the same boat. Last time I looked at the plans, I didn't > >>> see > >>> anything with low minimums. So even if you wrote code from scratch and > >>> never used it on more than 6 channels, you might have done something > >>> that normally requires a large upfront payment. Use $10k as an example. > >>> > >>> In such a case owner of the patent might have an attorney initiate > >>> contact. If you are willing to communicate they might allow you to pay > >>> the minimum and be licensed. If you can't do that, they might offer a > >>> settlement where you stop using the codec and pay them some lesser > >>> amount. > >>> > >>> If the patent holder can easily prove the violation you might as well > >>> try to deal with them and get things settled fast. If you sell or give > >>> away the codec it is easier for them to dig up proof. If you have > >>> unhappy employees that might be the way they hear about the > >>> violation in > >>> the first place. > >>> > >>> Important consideration: Bankruptcy law generally excludes debts > >>> created > >>> by things like malicious or criminal acts. > >>> > >>> Matthew Rubenstein wrote: > >>> > >>> > >>>> As far as I know, the g729 patent requires buying a license to > >>>> operate > >>>> any implementation of it, whether Digium's, Intel's, or any other. > >>>> Digium is set up to collect royalties (perhaps at a favorable rate) as > >>>> part of their license from the patent holder. I don't know about Intel > >>>> or any other. Or what the mechanics are for enforcing the patent on > >>>> someone who operates a codec without a license. > >>>> > >>>> >
Re: [asterisk-users] Some queries on g729 license.
All of which hassle and expense can be avoided by buying a license for Digium's codec, which is tested to work well with Asterisk (and might come with some support). And is pretty cheap per simul "call". I wonder whether that "per call" means "per codec instance", which could be multiple licenses on a single conference call, where multiple (even if not all) parties are getting de/encoded simultaneously. And whether there are other tools for editing (/mixing/transforming) g729 data, in realtime (streams) or not (files), and whether they require a license. Ideally sox or equivalent would work on g729, maybe with a codec plugin. On Mon, 2007-01-08 at 13:23 -0500, Paul wrote: > First point to tackle in any case involving patent, copyright or > trademark infringement is whether or not the infringing party would have > been qualified to buy any usage rights at all. In a case where you > license the Intel source(read the terms, it's not really that "free"), > you would be applying for a license under some plan that includes > certain minimum payments. Even if you wrote new source from scratch you > would be in the same boat. Last time I looked at the plans, I didn't see > anything with low minimums. So even if you wrote code from scratch and > never used it on more than 6 channels, you might have done something > that normally requires a large upfront payment. Use $10k as an example. > > In such a case owner of the patent might have an attorney initiate > contact. If you are willing to communicate they might allow you to pay > the minimum and be licensed. If you can't do that, they might offer a > settlement where you stop using the codec and pay them some lesser amount. > > If the patent holder can easily prove the violation you might as well > try to deal with them and get things settled fast. If you sell or give > away the codec it is easier for them to dig up proof. If you have > unhappy employees that might be the way they hear about the violation in > the first place. > > Important consideration: Bankruptcy law generally excludes debts created > by things like malicious or criminal acts. > > Matthew Rubenstein wrote: > > > As far as I know, the g729 patent requires buying a license to operate > >any implementation of it, whether Digium's, Intel's, or any other. > >Digium is set up to collect royalties (perhaps at a favorable rate) as > >part of their license from the patent holder. I don't know about Intel > >or any other. Or what the mechanics are for enforcing the patent on > >someone who operates a codec without a license. > > > > > >On Mon, 2007-01-08 at 10:51 -0500, Al Bochter wrote: > > > > > >>What about the free open source G729 > >> > >>Best regards, > >> > >>Al Bochter > >>Bochter Services > >>http://www.BochterServices.com/?t=Email > >> > >> > >> > >>Matthew Rubenstein wrote: > >> > >> > >> > >>> I connect to a PSTN carrier over SIP which requires me to connect with > >>>a g729 codec. I'm using them for just robocalling: Asterisk server > >>>originates calls which play a prerecorded file. Can I pre-encode those > >>>stored files in g729 so they don't need to be encoded for each call? If > >>>so, do I need a g729 license for each call, or just a license for the > >>>preencoder? If the robocalls accept incoming DTMF, do I need g729 > >>>licenses for those calls? > >>> > >>> > >>>On Mon, 2007-01-08 at 04:08 -0700, > >>>[EMAIL PROTECTED] wrote: > >>> > >>> > >>> > >>> > >>>>Date: Mon, 08 Jan 2007 13:47:39 +0800 > >>>>From: Leo Ann Boon <[EMAIL PROTECTED]> > >>>>Subject: Re: [asterisk-users] Some queries on g729 license. > >>>>To: Asterisk Users Mailing List - Non-Commercial Discussion > >>>> > >>>>Message-ID: <[EMAIL PROTECTED]> > >>>>Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >>>> > >>>>Xue Liangliang wrote: > >>>> > >>>> > >>>> > >>>> > >>>>>Hi, all > >>>>> > >>>>>I am a pabx vendor from Singapore. Recently we are going to > >>>>> > >>>>> > >>>>> > >>>>> > >>>>implement > >>>> > >>>> > >>>> > >>>> > >>>>>a failover solution for our customers using heartbeat, the asterisk > >>>>>server can failover perfectly, however the g729 codec canot work, > >>>>>because it is binded the mac address, we have bought two set of > >>>>>licenses, can you provide us some workaround for this scenario? > >>>>> > >>>>> > >>>>> > >>>>> > >>>>It shouldn't be a problem if you're only doing IP takeover and have > >>>>bound the licenses to each server separately. If you're sharing the > >>>>storage, then that could pose a problem. > >>>> > >>>>Leo > >>>>DatVoiz Singapore Pte Ltd > >>>> > >>>> > >>>> > >>>> > -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some queries on g729 license.
As far as I know, the g729 patent requires buying a license to operate any implementation of it, whether Digium's, Intel's, or any other. Digium is set up to collect royalties (perhaps at a favorable rate) as part of their license from the patent holder. I don't know about Intel or any other. Or what the mechanics are for enforcing the patent on someone who operates a codec without a license. On Mon, 2007-01-08 at 10:51 -0500, Al Bochter wrote: > What about the free open source G729 > > Best regards, > > Al Bochter > Bochter Services > http://www.BochterServices.com/?t=Email > > > > Matthew Rubenstein wrote: > > > I connect to a PSTN carrier over SIP which requires me to connect with > >a g729 codec. I'm using them for just robocalling: Asterisk server > >originates calls which play a prerecorded file. Can I pre-encode those > >stored files in g729 so they don't need to be encoded for each call? If > >so, do I need a g729 license for each call, or just a license for the > >preencoder? If the robocalls accept incoming DTMF, do I need g729 > >licenses for those calls? > > > > > >On Mon, 2007-01-08 at 04:08 -0700, > >[EMAIL PROTECTED] wrote: > > > > > >>Date: Mon, 08 Jan 2007 13:47:39 +0800 > >>From: Leo Ann Boon <[EMAIL PROTECTED]> > >>Subject: Re: [asterisk-users] Some queries on g729 license. > >>To: Asterisk Users Mailing List - Non-Commercial Discussion > >> > >>Message-ID: <[EMAIL PROTECTED]> > >>Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >> > >>Xue Liangliang wrote: > >> > >> > >>>Hi, all > >>> > >>>I am a pabx vendor from Singapore. Recently we are going to > >>> > >>> > >>implement > >> > >> > >>>a failover solution for our customers using heartbeat, the asterisk > >>>server can failover perfectly, however the g729 codec canot work, > >>>because it is binded the mac address, we have bought two set of > >>>licenses, can you provide us some workaround for this scenario? > >>> > >>> > >>It shouldn't be a problem if you're only doing IP takeover and have > >>bound the licenses to each server separately. If you're sharing the > >>storage, then that could pose a problem. > >> > >>Leo > >>DatVoiz Singapore Pte Ltd > >> > >> -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some queries on g729 license.
Thank you, that is excellent advice. I understand that Intel has a free g729 codec, and that there might be others. Free g729 codecs cheat Digium of some income that helps keep them producing free Asterisk (and hosting lists like this one), but what other reasons (quality, performance, missing features) would make the Digium (or other $) license worth paying for? On Mon, 2007-01-08 at 14:40 +, Thomas Kenyon wrote: > Matthew Rubenstein wrote: > > I connect to a PSTN carrier over SIP which requires me to connect with > > a g729 codec. I'm using them for just robocalling: Asterisk server > > originates calls which play a prerecorded file. Can I pre-encode those > > stored files in g729 so they don't need to be encoded for each call? > > Yes, if you are using asterisk 1.4 then in the CLI you can type: > > convert > . extension> .g729 > > so convert recording.ulaw recording.g729 > > Will make a permanent copy not requireing transcoding again. > > If you are using asterisk 1.2, there is a tool on the asteriskguru site > to transcode the file for you. > > http://www.asteriskguru.com/tools/audio_conversion.php > > > If > > so, do I need a g729 license for each call, or just a license for the > > preencoder? > > You will need a license for when the file is encoded, after that if it > is played back on a g729 call you will not need a license. Asterisk will > automatically choose the lowest cost file to playback (which one in > natvie format will be). > > > If the robocalls accept incoming DTMF, do I need g729 > > licenses for those calls? > > > > You only need a license when you are transcoding, if you have an > incoming call that is g729 and you terminate the call to a device that > is configured to use g729 then you will not need a license. > > If you are recording the call then you will need (possibly 2) llicenses. > > DTMF signals do not require a license (although the device generating > them needs to be configured to use RFC 2833 or Out of Band for DMTF > encoding). -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some queries on g729 license.
I connect to a PSTN carrier over SIP which requires me to connect with a g729 codec. I'm using them for just robocalling: Asterisk server originates calls which play a prerecorded file. Can I pre-encode those stored files in g729 so they don't need to be encoded for each call? If so, do I need a g729 license for each call, or just a license for the preencoder? If the robocalls accept incoming DTMF, do I need g729 licenses for those calls? On Mon, 2007-01-08 at 04:08 -0700, [EMAIL PROTECTED] wrote: > Date: Mon, 08 Jan 2007 13:47:39 +0800 > From: Leo Ann Boon <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Some queries on g729 license. > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Xue Liangliang wrote: > > Hi, all > > > > I am a pabx vendor from Singapore. Recently we are going to > implement > > a failover solution for our customers using heartbeat, the asterisk > > server can failover perfectly, however the g729 codec canot work, > > because it is binded the mac address, we have bought two set of > > licenses, can you provide us some workaround for this scenario? > It shouldn't be a problem if you're only doing IP takeover and have > bound the licenses to each server separately. If you're sharing the > storage, then that could pose a problem. > > Leo > DatVoiz Singapore Pte Ltd -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there any Asterisk controllable thermostat?
On Thu, 2006-12-07 at 07:20 -0700, [EMAIL PROTECTED] wrote: > Date: Thu, 07 Dec 2006 02:11:59 -0700 > From: John Marvin <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Is there any Asterisk controllable > thermostat? > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Doug Crompton wrote: > > I remembered I had an x10 bottlerocket in my X10 junkbox so I > connected it > > to a spare serial port on my linux server (asterisk resides there) > and > > implemented with some mods the code mentioned earlier > > > > > http://lorance.freeshell.org/asterisk/#asterisk-can-control-the-world > > > > and it works great. Now I have one more way to control X10 devices. > I can > > even call my VM on the way home and turn on my lights or whatever > before I > > get home. > > I would suggest that people who don't already have an investment in > home > automation equipment should look at Insteon rather than X10. Insteon > is > a next generation version of X10 that provides backwards > compatibility > with X10. The devices are a little more expensive, but not as > expensive > as some of the other alternatives. Insteon provides 2 way > communication > and is a lot more reliable than X10. > > If you already have an investment in X10 devices you can slowly > convert > to Insteon, since Insteon provides backwards compatibility, i.e. X10 > controllers can control Insteon devices and Insteon controllers can > control X10 devices, however you won't get all the advantages of > Insteon > until you have Insteon controllers controlling Insteon devices. > > For people with some soldering and basic circuit design skills, you > may > want to consider using ethernet as a home automation bus for some > things. I love the Olimex PIC WEB and PIC Mini Web development boards > (they cost $49.95 and $39.95 respectively). They have an ethernet > port > and an expansion connector for the available PIC I/O pins. Microchip > provides a free C compiler for Pic processors, and they also have an > open source networking stack that works on the Olimex boards. So with > a > ribbon cable connector and a small breadboard with a few IC's and/or > driver transistors you can build a device that responds to commands > via > the network (or via a built in web server) from your Asterisk server > that does about any task you can think of. Lots of fun ... I'm > currently > building a voicemail indicator (my wife didn't like me taking her > answering machine away with the blinking lights when we switched to > Asterisk voicemail) using a PIC Web board. Next project will be a web > based sprinkler controller. Are any of these home automation systems compatible with homeplug? Or WiFi, or BlueTooth? It seems to me that bundling a proprietary (or less popular) network protocol (and HW) with the device controller fragments the market, and prohibits reuse of the mass market network, which prevents economies of scale for consumers and developers. > John -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Standardized IVR UI Pattern (was: Re: [asterisk-users] Is there any Asterisk controllable thermostat?)
On Wed, 2006-12-06 at 23:51 -0700, [EMAIL PROTECTED] wrote: > Date: Wed, 06 Dec 2006 22:37:01 -0500 > From: Steve Prior <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Is there any Asterisk controllable > thermostat? > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Doug Crompton wrote: > > and it works great. Now I have one more way to control X10 devices. > I can > > even call my VM on the way home and turn on my lights or whatever > before I > > get home. > > > > Doug > > I've started to play with writing some code using the Java FastAGI > interface to connect to my home automation system. The code is > working and I could now write whatever I wanted, but I haven't figured > out what would be a reasonable menu interface that wouldn't be very > annoying to use. I'd be very interested to hear what menu structures > and what actual capabilities people have found useful and nice to use. > > For example, has anyone come up with something less annoying than the > following dialog: > > "Press 1 for living room, press 2 for outside, press 3 for bedroom" > (I press 2) > "Press 1 for porch light, press 2 for garage light" > (I press 1) > "Press 1 to turn on, Press 2 to turn off, Press 3 to say current > status" > (I press 1) > "congratulations, you just spent several minutes just to turn on a > light!" I don't know why IVR menus still include so much extra verbiage. They should act like numbered lists - everyone knows the stated number means the key to press, and the stated name means what you will get. So: (Listens for DTMF) Hello, this is home thermostat. 1 living room 2 outside 3 bedroom (waits for DTMF, maybe repeats after a 2 second pause) (I press 2) (Listens for DTMF) Outside 1 porch light 2 garage light (waits for DTMF, maybe repeats after a 2 second pause, offers to hangup after maybe 15 seconds) (I press 1) (Listens for DTMF) Outside Porch light 1 on 2 off 3 say current status (waits for DTMF, maybe repeats after a 2 second pause) (I press 1) (Listens for DTMF) Outside porch light status turned on star for options, hash to hangup (waits for DTMF, maybe repeats after a 2 second pause) That menu system would take about 10 seconds the first time through, listening to all prompts. Subsequent navigation could take 2-4 seconds. Subsequent shortcuts through a collapsed star-hash "menu" could take 1-2 seconds. Make the star key an "" key to the previous scope. Make the hash key an "" key that terminates any multiple-key entry. Collapse all menu scopes/items into a single long list that can be reached at any time through "star-hash". Introduce the whole menu system with "press star for options", to the star-star menu. Make the "0" option in the "star options" menu the path to a human operator, if there is one. And always immediately feedback to any received key with at least a click. This simple UI should be common to every IVR app, so anyone can always use it without listening for a while to learn how to navigate the IVR. In fact, I call this system "IKR" (Interactive Key Response), and maybe every system should answer the call with first saying "IKR". Then callers would immediately know when our skills on the common UI would work, without waiting to learn, or mistake it. If the server played a few touchtones, like "4-5-7" (keypad "IKR") while saying "IKR", smart automated clients could detect the system and use it. To complete the interactivity protocol, every spoken digit to be pressed in the numbered menus would also play the digits' DTMF. And the intro to the scope to which a client DTMF navigated would play the last digits that navigated there from the previous scope while saying the name of the new scope. This is the system that I used to use when I built dedicated IVR systems a dozen years ago (on Dialogic HW). Almost no IVR people were on the Internet then, before the Web. There was no community, and IVR vendors competed so harshly that they couldn't get such a standard interface going, even for mutual benefit. So now everyone hates using IVR, even when it's better than a human operator. And we still all roll our own from scratch. But with Asterisk, and web/maillists connecting a community, we can adopt a common system. If enough people like it, I will publish the spec, and maybe write the RFC. Or maybe there's a better one that will be adopted more widely more quickly, and we can get behind that. If you don't like it, you can still roll your own, just don'
RE: [asterisk-users] any possibility of Vonage Integration
On Wed, 2006-12-06 at 13:03 -0500, Vijay Gandhi wrote: > must say very nice & deep calcutaion Thank you. Did you test it for errors? There's also a factor of 6/6 (or whatever) billing vs Vonage $25/75 flat, which can save in generic bills. It might even save an average of about 10%, if calls average 5min, more/less for shorter/longer average calls. But again, any price savings competes with Vonage's simplicity, basic reliability, zero overhead costs, and support services, as well as other calling features and their include ongoing operational costs. > Regards > > Vijay Gandhi > > -----Original Message- > From: Matthew Rubenstein [mailto:[EMAIL PROTECTED] > Sent: Wednesday, December 06, 2006 12:29 PM > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] any possibility of Vonage Integration > > > On Wed, 2006-12-06 at 05:41 -0700, > [EMAIL PROTECTED] wrote: > > Date: Wed, 6 Dec 2006 12:21:12 +0200 > > From: "Dovid B" <[EMAIL PROTECTED]> > > Subject: Re: [asterisk-users] any possibility of Vonage Integration > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > > > Message-ID: <[EMAIL PROTECTED]> > > Content-Type: text/plain; format=flowed; charset="iso-8859-1"; > > reply-type=original > > > > > > > Don't be fooled by the flat rates of the locked-box providers. > > > The real rates are so low these days most people pay less paying > > > per minute than paying a Vonage style flat rate. In addition > > > people report if you start making really heavy usage of your > > > Vonage flat rate so that they are losing money on you, they notice > > > and try to stop it. > > > > > > $25/month will buy you close to 50 hours of urban SIP termination, > > > it's down to half a cent in some of the big cities. Are you > > > going to average 50 hours on the phone each month? Some people > > > do, but most don't. (Otherwise Vonage could not even pretend it is > > > going to make money.) > > <./snip> > > > > Like any other provider, look at Vonage's tos agreement. If you go > > over I > > believe 100 hours they slap you with a $50.00 fee. I have a provider > > that I > > pay $5.00 a month to for my did and they asked me to use up to 2-3 > > channels > > for incoming, however they never capped me. Once in a while I use up > > to 7-8 > > channels with no problem. ( I tested once with all the cell phones > > that I > > had and I got 10 channels at once !!). As for outbound I use voipjet > > which > > is 1.3 cents. Like it was said above if you do the math it may be > > worth it > > for you to drop vonage all together. > > I'm doing the math to find where Vonage and generic directly compete. > If someone can check it to find any typo noise that I amplified with > successive calculations, or other mistakes, I'd love to be corrected. > But even on pure minutes, Vonage looks better than generic in the "sweet > spots". > > At $0.01:minute for each leg of US48 termination with a generic brand > provider, $25:mo buys 41h:40m generic, or 20h:50m of 2-party calls > generic. 100h generic would cost $120. If Vonage charges $25 for up to > 99h:59m, that's already a savings of $94.99 (over 79% off). If Vonage > charges $50 penalty at 100h, that's $75 for 100h, still $45 off (37.5% > off). If that's the highest penalty threshold, then at the possible > maximum (31d*24h*60m = 44,640m or 744h) monthly minutes would cost > $892.80 generic, a maximum savings of $817.80 (over 91.5% off) at > Vonage. Average monthly 43,830m or 730h:30m is $876.60 generic, so $75 > Vonage save $801.60 (over 91% off). $24.99 buys 20h:49m generic, beating > Vonage; Vonage is always cheaper than generic above that duration. > > Cheaper @$0.02:min 2-party calls: > 00h:01m-20h:49m generic > 20h:50m+ Vonage > > > If minutes cost $0.005:minute per leg generic brand, $25 buys 41h:40m > generic, 99h:59m Vonage. 100h generic is $60, Vonage is $75, so Vonage > costs $15 more (125% of generic; generic is 20% off). $75 buys 125h > generic, but up to 744h Vonage (730h:30m average monthly). $74.99 buys > 124h:59m generic, but nothing more at Vonage than the 99h:59m that $25 > buys. So at that half-cent minute rate, 41h:39m and less costs less than > Vonage's minimum $25 (where $0.005 more buys you 99h59m). And generic is > cheaper than Vonage for total average monthly usage from > 100h:00m-124h:59m, from $0.01-$15 cheaper (from jus
Re: [asterisk-users] any possibility of Vonage Integration
05:min, or $27) each month. Or it's all just a stock scam to enrich Jeffrey Citron with another Bubble-type equity sale on a losing business, which a lot of people are saying. But the competition will still drive generic minutes rates lower, especially outside US48 where $0.01:min is rare, even shocking. -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there any Asterisk controllable thermostat?
On Mon, 2006-12-04 at 00:58 -0700, [EMAIL PROTECTED] wrote: > Date: Sun, 3 Dec 2006 23:04:52 -0500 > From: "Zeeshan Zakaria" <[EMAIL PROTECTED]> > Subject: [asterisk-users] Is there any Asterisk controllable > thermostat? > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > I am wondering if there is any such thermostat available which can be > controlled from Asterisk. Trixbox comes bundled with xPl, which is a home automation network API that is also common to Windows XP. I haven't seen any documentation of how to actually use it (with Trixbox/Asterisk), but I would be very interested in seeing some, including examples and supported HW. > Like you call your home pbx, dial some extension, > e.g. 333 and it asks to set the temperature, you enter a temperature, > and it > sets the thermostat to that temperature. This thermostat will be very > useful, e.g. when you're coming back home after a few days and now its > snowing and you want home to be warm on your arrival, you can turn the > furnace on an hour before your arrival. > > Is there any such thermostat available, and for that matter any other > Asterisk controllable home automation devices? > > -- > Zeeshan A Zakaria -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729 Passthru?
I have a SIP carrier which accepts only G729 connections from my Asterisk server. If all the server does is Dial() (out) two legs of a call which are natively bridged, with no processing the media (and no DTMF detection, etc), do I need to install a G729 codec of my own? All the media from each leg connected to the other is already encoded into G729 by the SIP carrier from which it's coming for feeding back to the SIP carrier. Does that "loopback" work without a G729 codec on the server? If not, what would the codec actually do with the data it gets? A related issue is whether I can pre-encode recorded audio files with a G729 codec. So the server can send "wakeup call" messages to the SIP carrier without running the codec at call time, just sending the pre-encoded media to the SIP carrier. -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel compilation problems with linux 2.6.19
On Sat, 2006-12-02 at 09:53 -0700, [EMAIL PROTECTED] wrote: > Date: Sat, 2 Dec 2006 11:51:37 +0200 > From: Tzafrir Cohen <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] zaptel compilation problems with linux > 2.6.19 > To: Asterisk-Users > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=us-ascii > > Hi Hi, and thanks for the help :). > On Fri, Dec 01, 2006 at 01:43:20AM -0500, Matthew Rubenstein wrote: > > On Thu, 2006-11-30 at 17:56 -0700, > > [EMAIL PROTECTED] wrote: > > > Message: 18 > > > Date: Fri, 1 Dec 2006 00:56:10 +0200 > > > From: Tzafrir Cohen <[EMAIL PROTECTED]> > > > Subject: Re: [asterisk-users] zaptel compilation problems with > linux > > > 2.6.19 > > > To: asterisk-users@lists.digium.com > > > Message-ID: <[EMAIL PROTECTED]> > > > Content-Type: text/plain; charset=us-ascii > > > > > > On Thu, Nov 30, 2006 at 02:44:03PM -0500, Matthew Rubenstein > wrote: > > > > I'm having problems installing ztdummy on my > > > > CentOS-2.6.16-rc6/Asterisk-1.2.11 datacenter server (no ZAP HW, > SIP > > > only > > > > to PSTN). I unpacked the kernel sources and headers in a > directory, > > > made > > > > (but not re/installed) the kernel, unpacked the zaptel-1.2.11 > > > tarball, > > > > then went thru the make sequence. It seemed to proceed OK > (without > > > > errors, just some warnings), but didn't seem to result in a > loadable > > > > ztdummy kernel module. Complete (failed) install session > transcript > > > is > > > > attached to this message; details appended: > > > > > > > > > > > > - > > > > # cd > > > > # export KSRC= > > > > # make clean > > > > # make config > > > > [... series of shell script conditionals apparently executed > OK ...] > > > > # make linux26 > > > > [... series of CC/LD reports, some warnings, no errors ...] > > > > # make install > > > > [... series of INSTALL messages, same warnings from (make > linux26), > > > no > > > > errors ...] > > > > # modprobe ztdummy > > > > FATAL: Module ztdummy not found. > > > > FATAL: Error running install command for ztdummy > > > > # modprobe zaptel > > > > FATAL: Module zaptel not found. > > > > > > > > - > > > > > > > > (make linux26) generated some warnings about various usb_*_dev > > > symbols > > > > undefined in [xpp,wcusb]/*.ko, but no actual errors. (make > install) > > > > > > Those are harmless, IIRC. I'll try to recall their source. > > > > I suspected as such. But I don't think the server has full > USB/UHCI > > support running, or fully installed: > > > > > - > > # lsmod > > Module Size Used by > > binfmt_misc12168 1 > > dm_mod 59512 0 > > thermal13864 0 > > processor 25284 1 thermal > > fan 4772 0 > > floppy 63172 0 > > generic 4836 0 [permanent] > > ide_generic 1504 0 [permanent] > > # modprobe usb_uhci > > FATAL: Module uhci_hcd not found. > > # modprobe uhci > > FATAL: Module uhci_hcd not found. > > > - > > > > > > > > repeated those warnings. (modprobe ztdummy) finished with > > > > > > Was depmod run? > > > > No, but trying it now (after the transcripted session) didn't > seem to > > help: > > > - > > # depmod > > # modprobe ztdummy > > FATAL: Module ztdummy not found. > > FATAL: Error running install command for ztdummy > > > - > > > > > > > uname -r > > > > # uname -r > > 2.6.16-rc6-060427a > > so depmod, modprobe and such will look > under /lib/modules/2.6.16-rc6-060427a , > but the modules were installed elsewhere: > > >
Re: [asterisk-users] zaptel compilation problems with linux 2.6.19
On Thu, 2006-11-30 at 17:56 -0700, [EMAIL PROTECTED] wrote: > Message: 18 > Date: Fri, 1 Dec 2006 00:56:10 +0200 > From: Tzafrir Cohen <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] zaptel compilation problems with linux > 2.6.19 > To: asterisk-users@lists.digium.com > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=us-ascii > > On Thu, Nov 30, 2006 at 02:44:03PM -0500, Matthew Rubenstein wrote: > > I'm having problems installing ztdummy on my > > CentOS-2.6.16-rc6/Asterisk-1.2.11 datacenter server (no ZAP HW, SIP > only > > to PSTN). I unpacked the kernel sources and headers in a directory, > made > > (but not re/installed) the kernel, unpacked the zaptel-1.2.11 > tarball, > > then went thru the make sequence. It seemed to proceed OK (without > > errors, just some warnings), but didn't seem to result in a loadable > > ztdummy kernel module. Complete (failed) install session transcript > is > > attached to this message; details appended: > > > > > - > > # cd > > # export KSRC= > > # make clean > > # make config > > [... series of shell script conditionals apparently executed OK ...] > > # make linux26 > > [... series of CC/LD reports, some warnings, no errors ...] > > # make install > > [... series of INSTALL messages, same warnings from (make linux26), > no > > errors ...] > > # modprobe ztdummy > > FATAL: Module ztdummy not found. > > FATAL: Error running install command for ztdummy > > # modprobe zaptel > > FATAL: Module zaptel not found. > > > - > > > > (make linux26) generated some warnings about various usb_*_dev > symbols > > undefined in [xpp,wcusb]/*.ko, but no actual errors. (make install) > > Those are harmless, IIRC. I'll try to recall their source. I suspected as such. But I don't think the server has full USB/UHCI support running, or fully installed: - # lsmod Module Size Used by binfmt_misc12168 1 dm_mod 59512 0 thermal13864 0 processor 25284 1 thermal fan 4772 0 floppy 63172 0 generic 4836 0 [permanent] ide_generic 1504 0 [permanent] # modprobe usb_uhci FATAL: Module uhci_hcd not found. # modprobe uhci FATAL: Module uhci_hcd not found. - > > repeated those warnings. (modprobe ztdummy) finished with > > Was depmod run? No, but trying it now (after the transcripted session) didn't seem to help: - # depmod # modprobe ztdummy FATAL: Module ztdummy not found. FATAL: Error running install command for ztdummy - > uname -r # uname -r 2.6.16-rc6-060427a > ls -l /lib/modules/2.6*/misc/*.ko # ls -l /lib/modules/2.6*/misc/*.ko -rw-r--r-- 1 root root 198617 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/pciradio.ko -rw-r--r-- 1 root root 195365 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/tor2.ko -rw-r--r-- 1 root root 122139 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/torisa.ko -rw-r--r-- 1 root root 114623 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/wcfxo.ko -rw-r--r-- 1 root root 164626 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/wct1xxp.ko -rw-r--r-- 1 root root 340812 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/wctdm24xxp.ko -rw-r--r-- 1 root root 215930 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/wctdm.ko -rw-r--r-- 1 root root 204323 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/wcte11xp.ko -rw-r--r-- 1 root root 155909 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/wcusb.ko -rw-r--r-- 1 root root 343208 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/zaptel.ko -rw-r--r-- 1 root root 106184 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/ztd-eth.ko -rw-r--r-- 1 root root 92153 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/ztd-loc.ko -rw-r--r-- 1 root root 72401 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/ztdummy.ko -rw-r--r-- 1 root root 98511 Nov 30 09:24 /lib/modules/2.6.16-rc6/misc/ztdynamic.ko > - > > Building /etc/modprobe.d/zaptel... > > *** > > *** WARNING: > > *** If you had custom settings in /etc/modprobe.d/zaptel, > > *** they have been moved to /etc/modprobe.d/zaptel.bak. > > CentOS? /etc/modprobe.d ? What version is it, exactly? I'm not sure which Cent
Re: [asterisk-users] zaptel compilation problems with linux 2.6.19
FATAL: Module zaptel not found. - On Thu, 2006-11-30 at 12:00 -0700, [EMAIL PROTECTED] wrote: > Date: Thu, 30 Nov 2006 19:19:14 +0200 > From: Roman Yeryomin <[EMAIL PROTECTED]> > Subject: [asterisk-users] zaptel compilation problems with linux > 2.6.19 > To: asterisk-users@lists.digium.com > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > Hello! > > I have problems compiling zaptel (tried 1.2.11, 1.2.10 and 1.4.0-beta2 > -- all > give the same error) with 2.6.19 kernel > > CC > [M] /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.o > In file included > from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xpd.h:26, > from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.c:28: > /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xdefs.h:93: > error: > conflicting types for 'bool' > include/linux/types.h:36: error: previous declaration of 'bool' was > here > In file included > from /home/roman/install/asterisk/zaptel-1.4.0-beta2/zconfig.h:9, > from /home/roman/install/asterisk/zaptel-1.4.0-beta2/zaptel.h:34, > from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xproto.h:29, > from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xpd.h:27, > from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.c:28: > include/linux/config.h:10:3: warning: no newline at end of file > make[3]: *** > [/home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.o] > Error 1 > make[2]: *** [/home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp] > Error 2 > make[1]: *** [_module_/home/roman/install/asterisk/zaptel-1.4.0-beta2] > Error 2 > make[1]: Leaving directory `/home/roman/install/kernel/linux-2.6.19' > make: *** [linux26] Error 2 > > seems that commenting out "typedef int bool;" in xpp/xdefs.h on line > 93 works > that out, but don't know if it's completely right thing to do > > Roman > -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel compilation problems with linux 2.6.19
FATAL: Module zaptel not found. - On Thu, 2006-11-30 at 12:00 -0700, [EMAIL PROTECTED] wrote: > Date: Thu, 30 Nov 2006 19:19:14 +0200 > From: Roman Yeryomin <[EMAIL PROTECTED]> > Subject: [asterisk-users] zaptel compilation problems with linux > 2.6.19 > To: asterisk-users@lists.digium.com > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > Hello! > > I have problems compiling zaptel (tried 1.2.11, 1.2.10 and 1.4.0-beta2 > -- all > give the same error) with 2.6.19 kernel > > CC > [M] /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.o > In file included > from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xpd.h:26, > from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.c:28: > /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xdefs.h:93: > error: > conflicting types for 'bool' > include/linux/types.h:36: error: previous declaration of 'bool' was > here > In file included > from /home/roman/install/asterisk/zaptel-1.4.0-beta2/zconfig.h:9, > from /home/roman/install/asterisk/zaptel-1.4.0-beta2/zaptel.h:34, > from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xproto.h:29, > from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xpd.h:27, > from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.c:28: > include/linux/config.h:10:3: warning: no newline at end of file > make[3]: *** > [/home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.o] > Error 1 > make[2]: *** [/home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp] > Error 2 > make[1]: *** [_module_/home/roman/install/asterisk/zaptel-1.4.0-beta2] > Error 2 > make[1]: Leaving directory `/home/roman/install/kernel/linux-2.6.19' > make: *** [linux26] Error 2 > > seems that commenting out "typedef int bool;" in xpp/xdefs.h on line > 93 works > that out, but don't know if it's completely right thing to do > > Roman > -- (C) Matthew Rubenstein # make clean rm -f torisatool makefw tor2fw.h radfw.h rm -f ztcfg torisatool makefw ztmonitor ztspeed zttool zttest fxotune rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f zonedata.lo tonezone.lo libtonezone.so *.lo make -C SUBDIRS= clean make[1]: Entering directory `' CLEAN /wct4xxp CLEAN /.tmp_versions make[1]: Leaving directory `' rm -rf .tmp_versions rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f fxotune rm -f core rm -f ztcfg-shared fxstest rm -rf misdn* rm -rf mISDNuser* # make config if [ -d /etc/rc.d/init.d ]; then \ install -D -m 755 zaptel.init /etc/rc.d/init.d/zaptel; \ //sbin/chkconfig --add zaptel; \ elif [ -d /etc/init.d ]; then \ install -D -m 755 zaptel.init /etc/init.d/zaptel; \ //sbin/chkconfig --add zaptel; \ fi if [ -d /etc/default ] && [ ! -f /etc/default/zaptel ]; then \ install -D -m 644 zaptel.sysconfig /etc/default/zaptel; \ fi if [ -d /etc/sysconfig ] && [ ! -f /etc/sysconfig/zaptel ]; then \ install -D -m 644 zaptel.sysconfig /etc/sysconfig/zaptel; \ fi if [ -d /etc/sysconfig/network-scripts ]; then \ install -D -m 755 ifup-hdlc /etc/sysconfig/network-scripts/ifup-hdlc; \ fi # make linux26 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits > tones.h cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\"makefw.c -o makefw ./makefw tormenta2.rbt tor2fw > tor2fw.h Loaded 69900 bytes from file ./makefw pciradio.rbt radfw > radfw.h Loaded 42096 bytes from file ZAPTELVERSION="1.2.11" build_tools/make_version_h > version.h.tmp if cmp -s version.h.tmp version.h ; then echo; else \ mv version.h.tmp version.h ; \ fi rm -f version.h.tmp cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o libtonezone.a -lm cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -O4 -g -Wall -DBUILDING_TONEZ
Re: [asterisk-users] zaptel compilation problems with linux 2.6.19
I'm having problems installing ztdummy on my CentOS-2.6.16-rc6/Asterisk-1.2.11 datacenter server (no ZAP HW, SIP only to PSTN). I unpacked the kernel sources and headers in a directory, made (but not re/installed) the kernel, unpacked the zaptel-1.2.11 tarball, then went thru the make sequence. It seemed to proceed OK (without errors, just some warnings), but didn't seem to result in a loadable ztdummy kernel module. Complete (failed) install session transcript is attached to this message; details appended: - # cd # export KSRC= # make clean # make config [... series of shell script conditionals apparently executed OK ...] # make linux26 [... series of CC/LD reports, some warnings, no errors ...] # make install [... series of INSTALL messages, same warnings from (make linux26), no errors ...] # modprobe ztdummy FATAL: Module ztdummy not found. FATAL: Error running install command for ztdummy # modprobe zaptel FATAL: Module zaptel not found. - (make linux26) generated some warnings about various usb_*_dev symbols undefined in [xpp,wcusb]/*.ko, but no actual errors. (make install) repeated those warnings. (modprobe ztdummy) finished with - Building /etc/modprobe.d/zaptel... *** *** WARNING: *** If you had custom settings in /etc/modprobe.d/zaptel, *** they have been moved to /etc/modprobe.d/zaptel.bak. [...] - but seemed to complete without errors. (make install) included a line - INSTALL /ztdummy.ko - Complete (failed) install session transcript is attached. On Thu, 2006-11-30 at 12:00 -0700, [EMAIL PROTECTED] wrote: > Date: Thu, 30 Nov 2006 19:19:14 +0200 > From: Roman Yeryomin <[EMAIL PROTECTED]> > Subject: [asterisk-users] zaptel compilation problems with linux > 2.6.19 > To: asterisk-users@lists.digium.com > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > Hello! > > I have problems compiling zaptel (tried 1.2.11, 1.2.10 and 1.4.0-beta2 > -- all > give the same error) with 2.6.19 kernel > > CC > [M] /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.o > In file included > from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xpd.h:26, > from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.c:28: > /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xdefs.h:93: > error: > conflicting types for 'bool' > include/linux/types.h:36: error: previous declaration of 'bool' was > here > In file included > from /home/roman/install/asterisk/zaptel-1.4.0-beta2/zconfig.h:9, > from /home/roman/install/asterisk/zaptel-1.4.0-beta2/zaptel.h:34, > from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xproto.h:29, > from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/xpd.h:27, > from /home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.c:28: > include/linux/config.h:10:3: warning: no newline at end of file > make[3]: *** > [/home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp/card_fxo.o] > Error 1 > make[2]: *** [/home/roman/install/asterisk/zaptel-1.4.0-beta2/xpp] > Error 2 > make[1]: *** [_module_/home/roman/install/asterisk/zaptel-1.4.0-beta2] > Error 2 > make[1]: Leaving directory `/home/roman/install/kernel/linux-2.6.19' > make: *** [linux26] Error 2 > > seems that commenting out "typedef int bool;" in xpp/xdefs.h on line > 93 works > that out, but don't know if it's completely right thing to do > > Roman > -- (C) Matthew Rubenstein # make clean rm -f torisatool makefw tor2fw.h radfw.h rm -f ztcfg torisatool makefw ztmonitor ztspeed zttool zttest fxotune rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f zonedata.lo tonezone.lo libtonezone.so *.lo make -C SUBDIRS= clean make[1]: Entering directory `' CLEAN /wct4xxp CLEAN /.tmp_versions make[1]: Leaving directory `' rm -rf .tmp_versions rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f fxotune rm -f core rm -f ztcfg-shared fxstest rm -rf misdn* rm -rf mISDNuser* # make config if [ -d /etc/rc.d/init.d ]; then \ install -D -m 755 zaptel.init /etc/rc.d/init.d/zaptel; \ //sbin/chkconfig --add zaptel; \ elif [ -d /etc/init.d ]; then \ install -D -m 755 zaptel.init /etc/init.d/zaptel; \ //sbin/chkconfig --add zaptel; \ fi if [ -d /etc/default ] && [ ! -f /etc/default/zaptel ]; then \ install -D -m 644 zaptel.sysconfig /etc/de
[asterisk-users] Dialout to Meetme Fails?
I'm trying to use Asterisk (v1.2.11) make a callback that dials both
legs of the call into a Meetme() room together, but I keep getting
"conf-invalid" messages.
I created a callfile (/var/spool/asterisk/outgoing/out.call) that
specifies a Local channel (extension) which contains a Dial() command to
the "dialer", and an extension which contains a Dial() command to the
"dialee". Each Dial() includes a G option to send the dialed terminal to
an extension which sends the dialed terminal's leg to a Meetme()
conference. The Dial completes, the G goes to the extension, but the
Meetme() seems to fail. I have an /etc/asterisk/meetme.conf with the
conf room defined. CLI> Meetme() returns "No valid conferences". How can
I check that Meetme() is installed/configured properly? What else could
be wrong?
/var/spool/asterisk/outgoing/out.call :
--
Channel: Local/[EMAIL PROTECTED]/n
Callerid: 646-750-82731
Context: ext-jjp-out
Extension: callTo
Priority: 1
Set: callFrom=12126661212
Set: callTo=12127773434
Set: callerID=2126661212
Set: sipCarrier=carrier
--
/etc/asterisk/meetme.conf :
--
[rooms]
conf => 9000
--
from /etc/asterisk/extensions.conf :
--
[ext-jjp-out]
{ ; HyCallBack
{ ; FROM
exten => callFrom,1,Noop(Calling SIP/${callFrom}@
${sipCarrier})
exten => callFrom,n,Dial(SIP/${callFrom}@
${sipCarrier},45,G(ext-jjp-out^conf^100)g)
exten => callFrom,n,Noop(Done dialing from)
} ; FROM
{ ; TO
exten => callTo,1,Noop(Calling SIP/${callTo}@
${sipCarrier})
exten => callTo,n,Dial(SIP/${callFrom}@
${sipCarrier},45,G(ext-jjp-out^conf^100)g)
exten => callTo,n,Noop(Done dialing to)
} ; TO
{ ; conf
exten => conf,100,Goto(ext-jjp-out,conf,150); dialer
landing
exten => conf,101,Goto(ext-jjp-out,conf,160); dialee
landing
exten => conf,150,Noop(dialer landing)
exten => conf,151,Goto(ext-jjp-out,conf,201); dialer
landing
exten => conf,160,Noop(dialee landing)
exten => conf,161,Goto(ext-jjp-out,conf,211); dialee
landing
exten => conf,201,Noop(dialee conferencing)
exten => conf,202,Meetme(9000)
exten => conf,203,Noop(dialee done conf)
exten => conf,211,Noop(dialer conferencing)
exten => conf,212,Meetme(9000)
exten => conf,213,Noop(dialer done conf)
} ; conf
} ; HyCallBack
--
--
(C) Matthew Rubenstein
___
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