Re: [asterisk-users] OT: What do you guys think of this?
Sounds possible, but as a user of uTorrent, I have yet to see this "feature" It may simply be that I havnt looked hard enough. I can say, that I still have to have a tcp port routed for uTorrent to work properly. I may post an update, If I notice a change in this behavour. --Christopher Dobbs On Mon, Dec 1, 2008 at 10:34 AM, Alex Balashov <[EMAIL PROTECTED]>wrote: > http://www.theregister.co.uk/2008/12/01/richard_bennett_utorrent_udp/ > > FUD? Interesting? Boring? New news? Old news? > > -- > Alex Balashov > Evariste Systems > Web: http://www.evaristesys.com/ > Tel: (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 > > ___ > -- 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 > -- find / -name "*base*" -user your -print | xargs 'chown us' ___ -- 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] iLBC and G729 codecs
I may be wrong about this, but * understads that these codecs exsist, but without a codec_XXX.so, it cant do translation on the codec. In this case, * can do pass throught (eg: g729 -> g729) but cannot do translation (eg: g729 -> gsm). You need to install the codec before you can do a translation. (g729 for one is somthing you must purchous a licensese for) --Chris On Sun, Sep 7, 2008 at 1:38 PM, Edgar Guadamuz <[EMAIL PROTECTED]> wrote: > Hi all, > > > In my modules.conf I have the autoload=yes, and there is one > codec_iLBC.so module in the modules folder. > > However, when I do show translation, I see no translation to/from iLBC > nor G.729, and I'm not able to establish call to channels using these > codecs. > > I read that there are some "formats" that can not be used as "codecs" > for live streams, but I actually didn't get it... Some light on this > will be apreciated! > > Thanks!! > > ___ > -- 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 > -- find / -name "*base*" -user your -print | xargs 'chown us' ___ -- 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] Microsoft CRM 4.0 integration with asterisk
I dont know if this will help, but I have been working with MS OCS at work, and * 1.6 integrates rather wall tith OCS speech server. If you need help on that relm, I can try to help. (admitidly I dont have inbound calls working, but we arnt worried about that, as our appplication is strictly outbound.) --Chris On Fri, Jul 11, 2008 at 2:16 AM, Jan Prunk <[EMAIL PROTECTED]> wrote: > Hello ! > > I am wondering if anyone has experiences with the integration of > Asterisk 1.4.19 into Microsoft Dynamics CRM 4.0 ? > Or alternatively integration with Microsoft Office Communications > server (however trying to avoid this, if it isn't really necessary for > the integration). > I would be glad to receive any links or manuals on this topic, which > helped you to integrate it. > > Kind regards, > Jan Prunk > -- > Jan Prunk > Website: http://www.prunk.si PGP key: 00E80E86 > Fingerprint: 77C5156E29A4EB6C1C4A5EBA414A29F500E80E86 > > ___ > -- 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 > -- find / -name "*base*" -user your -print | xargs 'chown us' ___ -- 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] Inbound Answer not working
Joseph L. Casale wrote: Attach to the Asterisk console and try making a call that usually fails with verbose set to 3 or so and post the output. It is probably something very simple. Your includes are probably the issue. Try bringing them all into extensions.conf and see if it works. Thanks, Steve Totaro Yup, I brought them in and its all working smooth now except there is a mis-configuration at my provider I am going to have to wait to get resolved. What about those includes was sketchy? Is that bad practice to separate them out? Thanks! jlc I use includes extensivly, and I provide phone service to several of my friends, the order you have them in can cause problems somtimes. That has been my experiance, for what its worth. -Chris ___ -- 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] RE: Asterisk RAS
(Sorry forgot to chaange the subject) Thankyou to all of you who replied. Seing as how this seemes to be an un-implemented feature under *, I will go ahead and write a handler for it. I will post here as I have progress. -Chris A side note: The reason I am doing this, is I do computer repair, I have brodband at my office, and have an * box doing my phones. I want to be able to "dial-in" with computers I am working on to test and make sure they are working, but I dont want to pay for a dialup account. I could order a line just for this, but it is something I do so rarly, that it would be more economical to just write a handler for *. ___ --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] call problem...
Thankyou to all of you who replied. Seing as how this seemes to be an un-implemented feature under *, I will go ahead and write a handler for it. I will post here as I have progress. -Chris A side note: The reason I am doing this, is I do computer repair, I have brodband at my office, and have an * box doing my phones. I want to be able to "dial-in" with computers I am working on to test and make sure they are working, but I dont want to pay for a dialup account. I could order a line just for this, but it is something I do so rarly, that it would be more economical to just write a handler for *. ___ --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] Can Asterisk RAS?
I am trying to set up somthing so I can dial into my asterisk box, and have it behave as a modem bank. Is there anything like that already, or am I going to have to write my own. I checked googls and found no leads, but thought I would ask here before I tried writing my own, just to make sure I wasnot reinventing the wheel. Thank you in advance for any responses. -Chris ___ --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] Equivalent of channel switching?
Barzilai wrote: I still haven't figured out what is the "best practices" or Asterisk-way to do traditional switching between channels in Asterisk. I come from traditional computer telephony where there are buses such as MVIP, with streams and timeslots. Asterisk, being born as a PBX solves most of the problems by "dialing a new extension". I'll present a ridiculous and hypothetical situation: User A has already dialed into Asterisk and is listening to some music through the phone. User B has already dialed into Asterisk and is listening to some weather forecast IVR. Event X happens (the Moon has aligned with Saturn) so we want A and B to start talking to each other. This doesn't involve DIALING a NEW extension, I guess... both of them are already inside the system. I can think of some convoluted way to do it; how do I do it the Asterisk way? BarZ ___ --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 If you transfered them both to a meetme, they could both talk, meetme can also dynamicly create the room, so you could try to use that. Christopher Dobbs ___ --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] Soekris and Asterisk
trixter http://www.0xdecafbad.com wrote: On Wed, 2005-10-12 at 17:46 -0700, Paul Mahler wrote: You need about 30MHz per channel. That means the Soekris can only handle part of a T1, it will never handle a quad span. Paul How was that determined? I have a problem with a plain number like that, which may have been taken into account, why I am asking... Different cpus operate differently, taking more or less time to complete certain functions. Instruction optimization can go a long way if those instructions are used (not terribly likely if its just pushing bits but there are some for just that). Additionally there is no codec processing (presumably) with TDMoE, does the 30MHz take into account any codec processing or is it literally 30MHz (on what cpu class?!) for just pushing bits? There are other factors, but you did say 'about' so they are optional to this conversation, ie other IRQs on the box, potential for device polling, etc. A tuned system for that specific task (pushing bits between a TDM card and ethernet via TDMoE) may be able to operate at a lower clock speed per channel, but that isnt as important for the initial questions. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users MHZ is not a valid way of gauging performance. It's all about the MIPS (Millions of Instrictions Per Second), Baby :). I was testing with some of the Soekris boards about a year ago for an client, the need was to make a TDMoE -> TDMoE router for a wireless network. (Yes I know that that is a stupid idea, and I told the client that it was a waist of his money to have me try.) the board I was using I think was the 4801, not sure thoe (It was a year ago) but it would pust 48 TDMoE channels at once over 100BaseT ok. So I would think that It would. I was using a customized linux distro, (as in one I created) contact me off list if you would like a copy of the distro. -- Christopher Dobbs Wireless Administrator Valario Inovations ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on windows
Matt wrote: Extremely good point... I myself am a Linux person, but manage several Windows machines (several meaning 25 or so). There is definately a time and place for Windows.. I'm just not sure a real-time-VoIP server is the time or place.Being semi-half serious about the GUI there also.You install X on your Asterisk server and things will not be happy either. I Run SuSE 9.3 with KDE 3.4, Asterisk 1.0.3, play MP3's and OGG's, SAMBA services, HTTPD, VNC, MicroWindows, FTP, SMTP, POP, IMAP, plus others. I dont see that the GUI slows things down to much, unless I am running a test and gring the call volume over 500 active calls. (I am developing a new channel driver for * ment for inclusion in mobile phones, think Asterisk+Cell Phone). The assertion that a GUI will bring a system to it's knee's is utter CRAP! It all has to do whith what the system is doing besides, and what the hardware can handle. BTW: the system this all is running on is an AMD 1700+, and the same system that I am using to brows the mailing list. --Christopher Dobbs --I think I think, There for I think I am. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wireless LANs and Asterisk
We are deploying an * soultion at the WISP that I freelance for. We are using a tranport that I designed called MATE: Multplexed Audio Transmited over Ethernet MATE is designed to be a better TDMoE. It uses uLAW and huffman compression. We also use custom Customer Premis Equipment that garenties the dilivery of the MATE streams. So far MATE supports 64 channels per stream. Streams are MAC -> MAC. Each MATE client note can support 16 streams. Each MATE server node can suport unlimmited streams. The system is still in prototype stage, but will be in full use within about three months. -- Christopher Dobbs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Direct MP3 channel Black Hole?
We do exactly that :) exten ,1,MusicOnHold(KissMyA$$) -- Christopher Dobbs Puddle wrote: I'm curious is it possible to direct a call to an extension that takes you straight to music on hold, but NOT the standard music on hold. The boss suggested something he wondered if it was possible. Example: Someone calls (Telemarketer), we answer tell them to hold while we 'redirect' them to extension (Someone Important) 666 which is a separate music on hold pool of mp3's from each of the employee's stating why we hate telemarketers. We think it's a good 'company' relief any help or suggestions greatly appreciated. -William __ Do you Yahoo!? Yahoo! Mail - 250MB free storage. Do more. Manage less. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXTEL errors !
Use FWDNET.NET. It is far better on call quality!! -- Christopher Dobbs Manjit Riat wrote: I am testing IAXTEL and routing 800 number to them.. Sometimes the call goes through and the other times it get the following error. WARNING[20502]: chan_iax2.c:1477 attempt_transmit: Max retries exceeded to host 69.73.19.178 on IAX2/iaxtel/3 (type = 6, subclass = 9, ts=631, seqno=1) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.1 - Release Date: 1/19/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] automatic startup
I know that several people have sent suggestions about init scripts, but, I use inittab to re spawn * sot to do a restart all I do is issue a "Stop Now" at the CLI. I am including my scripts and an excerpt of some of my system files. All tho, the suggestion about loading the modules in rc.local is pure gold. -- Christopher Dobbs Michael Graves wrote: Hi All, I've been thinking about taking steps to make my * server more reliable. In particular I'd like to have it automatically start * after a power loss. Can anyone here provide some guidance as to how to accomplish this. Keep in mind that I have a TDM400p that needs a couple of modprobe commands before I can start * itself. I had a look at the wiki but it seems light on this topic. Thanks, Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users #! /bin/bash chvt 24 asterisk -vvc &> /dev/tty24 < /dev/tty24 sleep 5 #! /bin/bash asterisk -r ast:12345:respawn:/sbin/astmain pbx:x:0:0::/root:/sbin/astrun #! /bin/bash exec ssh [EMAIL PROTECTED] #! /bin/bash exec ssh [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to connect two Asterisks as secure as possiblewithout too much additional bandwidth ?
This problem is being solved. See http://lists.digium.com/pipermail/asterisk-users/2004-November/073666.html I am currently in pre-testing phase of development. Features include: Optional Secondary Compression Selectable Encryption Level, from 32bit to 1024bit Uses UDP Voice and Data over same Link Trunking ADSI Support -- Christopher Dobbs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do I need to build up DID services?
As far as I know, Asterisk/Zaptel does not support analog DID service. --Eric I am thinking that there is some confusion about DID's. The only important difference between an analog DID and a POTS line is that when you pick up a DID, you are sent via DTMF the number (or a portion there of) that is being called. For incoming calls, set immediate=yes For outgoing calls Dial(Zap/Gx||D(12345)) This works. I am using this to interface an * box between a set of DID's and a radio phone patch. Hope this helps. -- Christopher Dobbs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Qestion about TDM over enthernet
The WiKi can give you step by step instructions, but I have had only failure with TDMoE. -- Christopher Dobbs FCG ZHAO Zigang wrote: >who can tell me how to do TDM over enthernet ? > >pc a connect pc b only use TDM card? > >thank you > >John. > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] polycom and cdp
Sorry, Replied to wrong message:) -- Christopher Dobbs Christopher Dobbs wrote: Here is a list of the libs I am using ld-2.2.5.so* ld-linux.so.2@ libc-2.2.5.so* libc.so.6@ libcom_err.so.2@ libcom_err.so.2.0* libcrypt-2.2.5.so* libcrypt.so.1@ libdl-2.2.5.so* libdl.so.2@ libe2p.so.2@ libe2p.so.2.3* libext2fs.so.2@ libext2fs.so.2.4* libm-2.2.5.so* libm.so.6@ libncurses.so.5@ libncurses.so.5.2* libnsl-2.2.5.so* libnsl.so.1@ libnss_dns-2.2.5.so* libnss_dns.so.2@ libnss_files-2.2.5.so* libnss_files.so.2@ libproc.so.2.0.7* libresolv-2.2.5.so* libresolv.so.2@ libutil-2.2.5.so* libutil.so.1@ [EMAIL PROTECTED] libcrypto.so@ libcrypto.so.0@ libcrypto.so.0.9.6* libss.so@ libssl.a libssl.so@ libssl.so.0@ libssl.so.0.9.6* libwrap.a libwrap.so.0@ libwrap.so.0.7.6* libz.so@ libz.so.1@ libz.so.1.1.4* libcrypto.a libcrypto.so@ libcrypto.so.0@ libcrypto.so.0.9.6* libss.so@ libssl.a libssl.so@ libssl.so.0@ libssl.so.0.9.6* libwrap.a libwrap.so.0@ libwrap.so.0.7.6* libz.so@ libz.so.1@ libz.so.1.1.4* -- Christopher Dobbs Richard wrote: Hi, Has anyone tried to use cdp to push the voice vlan tag to polycom phones? The document says that it is supported, but I can't make it work. Thanks, Richard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] polycom and cdp
Here is a list of the libs I am using ld-2.2.5.so* ld-linux.so.2@ libc-2.2.5.so* libc.so.6@ libcom_err.so.2@ libcom_err.so.2.0* libcrypt-2.2.5.so* libcrypt.so.1@ libdl-2.2.5.so* libdl.so.2@ libe2p.so.2@ libe2p.so.2.3* libext2fs.so.2@ libext2fs.so.2.4* libm-2.2.5.so* libm.so.6@ libncurses.so.5@ libncurses.so.5.2* libnsl-2.2.5.so* libnsl.so.1@ libnss_dns-2.2.5.so* libnss_dns.so.2@ libnss_files-2.2.5.so* libnss_files.so.2@ libproc.so.2.0.7* libresolv-2.2.5.so* libresolv.so.2@ libutil-2.2.5.so* libutil.so.1@ [EMAIL PROTECTED] libcrypto.so@ libcrypto.so.0@ libcrypto.so.0.9.6* libss.so@ libssl.a libssl.so@ libssl.so.0@ libssl.so.0.9.6* libwrap.a libwrap.so.0@ libwrap.so.0.7.6* libz.so@ libz.so.1@ libz.so.1.1.4* libcrypto.a libcrypto.so@ libcrypto.so.0@ libcrypto.so.0.9.6* libss.so@ libssl.a libssl.so@ libssl.so.0@ libssl.so.0.9.6* libwrap.a libwrap.so.0@ libwrap.so.0.7.6* libz.so@ libz.so.1@ libz.so.1.1.4* -- Christopher Dobbs Richard wrote: Hi, Has anyone tried to use cdp to push the voice vlan tag to polycom phones? The document says that it is supported, but I can't make it work. Thanks, Richard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem ringing simultaneous channels
FXS ports are always answered immediately. Do you have to dial out over PTSN? If so you are going to have an interesting time breaking this problem. -- Christopher Dobbs Russell Horn wrote: Alexander, I'm afraid it's POTS (X101P) and from what I have seen since I posted this is my problem. I wouldn't mind it hanging up the IAX2 channel and then calling it again, but it seems that everytime the new call via Zap/2 means no other calls are possible. There is ISDN in the office, but I don't have any access until April :/ If what I'm trying is impossible it will just have to wait Russell. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] gumstix
I have an embedded Linux Distro that is specifically designed to fit on as small as a 32MB CF card. Includes: HTTP Server SSH Client and server DHCPCD DHCPD bash and more Contact me off list if you are interested. -- Christopher Dobbs Michael Graves wrote: On Wed, 22 Dec 2004 12:39:46 -0600, Kristian Kielhofner wrote: Michael Graves wrote: They look cute, but not enough RAM for *. Someone already has * ported to the Soekris 4801. Have a look at www.soekris.com. Michael I didn't have to "port" Asterisk, the Soekris boards have 586's on them, I just compiled Asterisk accordingly and copied the binaries. Easy. As far as RAM, I am using Asterisk on a PC Engines WRAP with "only" 64mb of RAM, and it works fine. You can remove modules, tune some Makefile vars to make * run smaller (I didn't have to for 64mb), and that's without tweaking the code. I would love to see someone get * to run on a gumstix. Why I don't know, but how does "coolness" sound? My appologies. I thought the 64 MB was too small for actual production use. I'm hoping to change my * server to booting from an IDE flash module over the holidays. I agree that the Soekris and Gumstix platforms are cool, but I've already got a working server. I just want it to behave more like an appliance. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A few simple (I hope) questions from a first-timer
There is a better way, have you looked at RxFAX and TxFAX applications. I my,self have not used them, but there is plenty of info in the WiKi. Asterisk will branch to a fax extension after an answer() if you have it set up that way. -- Christopher Dobbs Joel Moore wrote: We have the following situation: We're ordering two PSTN lines for our new office (no broadband at all -- it's not even available). The first line is going to be our primary # and it will also serve as our fax line using distinctive ring. The second line is going to be for a dial-up ISP (*sigh*) but now we'd like it to do double duty as the second line of a hunt group (we can't even get voicemail service for these lines from Verizon). We'd be willing to hamper our Internet access a little to make this work. We'll be ordering a 2 FXO, 2 FXS TDM400P card for this setup. Now the questions: 1) I want to make sure I understand Asterisk's support for distinctive ring properly. I assume I can branch the first line into our fax machine before I plug it into the * server and that * can be configured to ignore the fax line's distinctive ring. Correct? Is there a better way to do this? Would we be better off letting * handle faxes? Is there anyway to pair up * with some fax software and let the * PC be the fax machine (thereby enjoying some additional benefits such as routing faxes to an email server)? 2) Is there any way to use * with the data line (i.e. does it have some sort of PPP module?) or do I have to branch that line into a modem before plugging it into the TDM400P card? 3) Does call waiting work with *? I see some messages saying that call waiting is a wasted feature but the Asterisk web site mentions call waiting is supported. Despite have 2 lines on the hunt group we'd like to also have call waiting so we can answer incoming calls when boths lines are tied up. Also, An ideal situation would be where the data line gets put on hold (something similar to the ICN feature found on USR modems) whenever the second line is needed for voice (incoming or outgoing). This may be too complicated, though. Joel Moore ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Startup Scripts (My Bad)
Sorry to all. I sent the wrong shell scripts. Here are the correct ones. Again, sorry to all! -- Christopher Dobbs Shahed wrote: Hi Christopher, The files you posted for inittab have a reference to astmain. I cant find this file anywhere (including WiKi / google etc). Is this a custom script that you have written ? Regards Shahed #! /bin/bash chvt 24 asterisk -vvc &> /dev/tty24 < /dev/tty24 sleep 5 #! /bin/bash asterisk -r ast:12345:respawn:/sbin/astmain pbx:x:0:0::/root:/sbin/astrun #! /bin/bash exec ssh [EMAIL PROTECTED] #! /bin/bash exec ssh [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why does * only work with an ancient mpg123?
I have: High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3. Version 0.59s-mh4 (2000/Oct/27). Written and copyrights by Michael Hipp. It works verry Well. -- Christopher Dobbs Remco Barende wrote: On Mon, 20 Dec 2004, Eric Wieling aka ManxPower wrote: Remco Barende wrote: Hi list! Just wondering, why is * sticking with an mpg123 version from the stoneage? Gentoo comes with 0.59s-r8 and this version doesn't even start. Ik know I could forcibly unmerge mpg123 and install the old version but I guess some day newer versions will have to be supported? Asterisk sets the following mpg123 options: "mpg123 -q -s --mono -r 8000 -b 2048 -f 4096" -q, --quiet Quiet. Suppress diagnostic messages. -s, --stdout The decoded audio samples are written to standard output, instead of playing them through the audio device. This option must be used if your audio hardware is not supported by mpg123. The output format is raw (headerless) linear PCM audio data, 16 bit, stereo, host byte order. -r rate, --rate rate Set sample rate (default: automatic). You may want to change this if you need a constant bitrate independed of the mpeg stream rate. mpg123 automagically converts the rate. You should then combine this with --stereo or --mono. -b size, --buffer size Use an audio output buffer of size Kbytes. This is useful to bypass short periods of heavy system activity, which would normally cause the audio output to be interrupted. You should specify a buffer size of at least 1024 (i.e. 1 Mb, which equals about 6 seconds of audio data) or more; less than about 300 does not make much sense. The default is 0, which turns buffering off. -f factor, --scale factor Change scale factor (default: 32768). Pretty much any program that accepts these options to generate raw (headerless) linear PCM audio data, 16 bit, mono, host byte order, at 8khz to stdout will work. At this time the only one that does this that I know is mpg123 0.59r Thanks! But when I look at the output of mpg123 0.59s-r8 all these commandline switches are still supported, why it it only the old version that is supported, not the newer ones? asterisk # mpg123 High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3. Version 0.59s-r8 (2000/Oct/27). Written and copyrights by Michael Hipp. Uses code from various people. See 'README' for more! THIS SOFTWARE COMES WITH ABSOLUTELY NO WARRANTY! USE AT YOUR OWN RISK! usage: mpg123 [option(s)] [file(s) | URL(s) | -] supported options [defaults in brackets]: -vincrease verbosity level -qquiet (don't print title) -ttestmode (no output) -swrite to stdout -w write Output as WAV file -k n skip first n frames [0]-n n decode only n frames [all] -ccheck range violations -yDISABLE resync on errors -b n output buffer: n Kbytes [0]-f n change scalefactor [32768] -r n set/force samplerate [auto]-g n set audio hardware output gain -os,-ol,-oh output to built-in speaker,line-out connector,headphones -a d set audio device -2downsample 1:2 (22 kHz)-4downsample 1:4 (11 kHz) -d n play every n'th frame only -h n play every frame n times -0decode channel 0 (left) only -1decode channel 1 (right) only -mmix both channels (mono) -p p use HTTP proxy p [$HTTP_PROXY] -@ f read filenames/URLs from f -zshuffle play (with wildcards) -Zrandom play -u a HTTP authentication string -E f Equalizer, data from file See the manpage mpg123(1) or call mpg123 with --longhelp for more information. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Q about IAX (and IAXy)
As Per The WiKi: IAX sends both command information and voice data over the same connection. This allows it th transverse a NAT seamlessly. As for Double NAT, My setup is: Home PBX <[Wireless]> ISP WiFi NAT <[Ethernet]> Primary NAT <[Ethernet]> Work PBX So, Yes it will work over double NAT, I can send and receive calls at home. In fact, I call my wife all the time from work. -- Christopher Dobbs Nabeel Jafferali wrote: This is somewhat related to my other query on the list regarding NAT traversal. I have heard many times that IAX is "NAT-transperant". I am unsure how it accomplishes this. I do know that SIP works like this: your SIP device send a request to the SIP server (usually on port 5060) with whatever command. The SIP server respends to your device's "apparent" IP and port (this is decided depending on how that NAT is set up, STUN, etc.). The voice is then sent to the "apparent" RTP port on your device (deciding what that is, again, would depend on the NAT set up). How does IAX eliminate this problem of ports being "mapped" by your NAT router and external IPs? Does it use one port for both commands and voice packets? Does the remote server just use the "received from" IP address and port to respond? Finally, would an IAXy work seamlessly in a configuration where it is plugged into a NAT router which is plugged into another NAT router - double NATted? The * server is on a public IP. -- Nabeel Jafferali tel: 647.722.8457 x201 718.606.4190 x201 fwd: 46990 x201 email/msn: nabeeljafferali.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disabling " !" command
--> Sorry, Forgot to attach the files!! :) I decided, since there was interest, to send them to the who list. Here they are. BTW: I am also sending an excerpt from my passwd and inittab files. the passwd entry is so that i can ssh into [EMAIL PROTECTED] and get an asterisk CLI. This "Feature" is why I have "!" disabled. I hope that this helps people out. I would strongly suggest this only if: 1) The "!" command is disabled int the CLI. 2) Telnet is COMPLETELY disabled!!! 3) SSH access is only by the use of RSA keys. This is the setup that we use. It works very good for us. -- Christopher Dobbs ast:12345:respawn:/sbin/astmain pbx:x:0:0::/root:/sbin/astrun #! /bin/bash exec ssh [EMAIL PROTECTED] #! /bin/bash exec ssh [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disabling " !" command
I decided, since there was interest, to send them to the who list. Here they are. BTW: I am also sending an excerpt from my passwd and inittab files. the passwd entry is so that i can ssh into [EMAIL PROTECTED] and get an asterisk CLI. This "Feature" is why I have "!" disabled. I hope that this helps people out. I would strongly suggest this only if: 1) The "!" command is disabled int the CLI. 2) Telnet is COMPLETELY disabled!!! 3) SSH access is only by the use of RSA keys. This is the setup that we use. It works very good for us. -- Christopher Dobbs ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disabling " !" command
I have asterisk auto-magicly start from inittab. I have yest to try something that can do anything to the server (other than stop asterisk) from the * CLI. If anyone would like a copy of the scripts i use, contact me off-list. BTW: I had already removed the ! command from * before using it this way. -- Christopher Dobbs Roy Sigurd Karlsbakk wrote: since I run asterisk as root with a CLI open on TTY12 I was wondering if the "!" (shell) command can be disabled from the config, for safety reasons it seems me usefully. well. IMHO if someone can get access to your asterisk console, they can always ctrl+z or shutdown now or something. secure your server. don't trust asterisk to do it roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk Prompts
Thank you!! -- Christopher Dobbs Brian Wilkins wrote: All, Enjoy these free prompts as an addition to your sounds collection. I hope you find them useful. You can find them attached to this message. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Hardware
What codec is your soft phone using? Some of the codecs stink, also is the link to the * server heavily used? -- Christopher Dobbs Nihal wrote: Does some hardware just not work very well with Asterisk? I've got a fresh installation on a Fedora C2, P4x2, 2GB Ram. While listening to the demo over a softphone (over the LAN) I get a number of crackles and skips. IS THIS NORMAL FOR ASTERISK? Or is it hardware related? Thanks, Nihal ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call on hold disconnects...
Are you having the phone place the person on hold, or are you having * place them on hold? I dial #700 and it puts them on hold and they stay there, it also reads off to me the number I dial to get them off hold. REF: /etc/asterisk/features.conf -- Christopher Dobbs Shoval Tomer wrote: That's both true and false. We have a legacy PBX here. Panasonic make. Analog extensions connected to it (a.k.a "stupid" extensions) behace exactly like the grandstream - you can put a call on hold, but if you put the handset back on the cradle it's bye bye Mary. Digital extensions (a.k.a "smart" extensions) can hold a call indefinitely. They can do other neat stuff too... -Original Message- From: Ferguson, Michael [mailto:[EMAIL PROTECTED]] Sent: Friday, December 17, 2004 11:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Call on hold disconnects... Antony, Thanks. It seems that the GS will not keep the call on hold. In the real world though, when you place a call on hold, it is held until further action. The caller will hear messages, music, anything while you are gone to look for a file, etc. Technically, if you place the call on hold and put the handset back on the cradle, you DID NOT HANG UP to end the call. If you want to hang up the call you will first have to take the call off hold... No. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED]] On Behalf Of Antony Stone Sent: Friday, December 17, 2004 3:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Call on hold disconnects... On Friday 17 December 2004 20:43, Ferguson, Michael wrote: OK. I guess I was not clear. Sorry. The phone rings. The person picks up the handset and speaks to the caller. He then puts the call on hold by pressing the "HOLD" button on the GS 100 phone. The caller hears music on hold. So far, so good. The hand set is placed back on the cradle (as is done on a regular phone with a hold button) I'm not sure I agree with this. Some phones may allow you to hang up and not disconnect the call, but I don't think it's universal. Some phones interpret this to mean "oh, you want to hang up? Okay - I'll hang up the call then." The call is disconnected. Well, yes, because you hung up. What happens if you do something else, like dial another extension, or press the hold button again (perhaps to retreive the original caller)? I repeat one of my original questions - if this is not what you expected to happen when you hang up the phone, how would you expect to hang up the call when you wanted to? Antony. -- This email is intended for the use of the individual addressee(s) named above and may contain information that is confidential, privileged or unsuitable for overly sensitive persons with low self-esteem, no sense of humour, or irrational religious beliefs. If you have received this email in error, you are required to shred it immediately, add some nutmeg, three egg whites and a dessertspoonful of caster sugar. Whisk until soft peaks form, then place in a warm oven for 40 minutes. Remove promptly and let stand for 2 hours before adding some decorative kiwi fruit and cream. Then notify me immediately by return email and eat the original message. Please reply to the list; please don't CC me. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail without prompt
The way I have it set up, is that the mailbox is the same as the exten. I then wrote a macro that does it for me. [macro-stdiax] ; ARG1 = User ; ARG2 = Voice Mail Number exten => s,1,Dial(IAX2/${ARG1}/[EMAIL PROTECTED]||Ttr) ;exten => s,2,Voicemail(u${ARG2}) ;exten => s,3,Hangup ;exten => s,102,Voicemail(b${ARG2}) ;exten => s,103,Hangup macro-eracewcustomer] ; ARG1 = User exten => s,1,Dial(IAX2/${ARG1}/[EMAIL PROTECTED]|20|Ttr) exten => s,2,Voicemail(u${ARG1}) exten => s,3,Hangup exten => s,102,Voicemail(b${ARG1}) exten => s,103,Hangup [macro-stdtrunk] ; ARG1 = Zap Port ; ARG2 = Voice Mail Number exten => s,1,Dial(Zap/${ARG1}|20|Ttr) exten => s,2,Voicemail(u${ARG2}) exten => s,3,Hangup exten => s,102,Voicemail(b${ARG2}) exten => s,103,Hangup [macro-dialtrunk] exten => s,1,Dial(${TRUNK}/${ARG1}) exten => s,2,Congestion [macro-dialiax] exten => s,1,Dial(IAX/${ARG1}/${ARG2}) exten => s,2,Congestion [dialing_context] exten => 2050,1,Macro(stdtrunk,1,${EXTEN}) exten => 0205,1,Macro(stdiax,${EXTEN},${EXTEN}) -- Christopher Dobbs Antony Stone wrote: On Friday 17 December 2004 21:25, Ross Kevlin wrote: this would still only work if the mailbox number was the same as the caller id. I need some way to get the actual mailbox number of the caller. Where / how are your mailbox numbers stored? It shouldn't be too difficult to create a script or DB request to provide the CID and get the mailbox number in response? Just out of interest, why don't you make the mailbox ID = caller ID? Antony. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ethernet Channel Bank (Comming Soon to a NOCNear You!)
We have discussed T1/E1 Modules for our channel banks, but, our main focus at this time is FXO/FXS interfaces. A T1 card is about $440 US. An Ethernet card is about $15 US. That is why we are doing this. Further more, we believe that we can sell these units cheaper than a T1/E1 channel bank, increasing the savings to the end user. (P.S. If Mr Spencer would like to comment on what Digium would think of these units, I Would appreciate it.) -- Christopher Dobbs Matthew Boehm wrote: And T1s too. If you can supply a seperate piece of hardware that can handle all the T1 crap then pass calls to asterisk as SIP/IAX, that would be awesome in our situation. Right now our only solution is 8 T1s into a 5300, then SIP to asterisk. -Matthew - Original Message - From: "Marc Storck" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Monday, December 13, 2004 8:09 PM Subject: Re: [Asterisk-Users] Ethernet Channel Bank (Comming Soon to a NOCNear You!) Will you have a Channel Bank for E1, many E1s instead of the FXS/FXO ports?? Marc Christopher Dobbs wrote: My company has started development on a Ethernet based channel bank. Here are the (current) spec's - 10/100 Ethernet Port - Up to 96 FXS/FXO ports (Thats 4 DS1's for the math impaired) - Serial Console - TDMoE - IAX2 - EETP (A protocol that we have designed for IP Telephony) We have just started prototyping this device, so... -- Christopher Dobbs ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- CTOMarc Storck MS Networks SA [EMAIL PROTECTED] Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352 2727 3060 -- LuxAdmin powered service --- http://www.Gateway.lu Your gateway to the net Advantages of ADSL solutions by LuxAdmin: - price: cheap and clear - products: proven quality - support: friendly and helpful --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ethernet Channel Bank (Comming Soon to a NOC Near You!)
Had not thought of that. We are mostly interested in providing POTS lines inside of a company. Think of it as a a massive CPE device running on the VPN of a corporation with offices in many towns. who knows? We saw all of the traffic about Ethernet channel banks and started talking, and well, we are going to try this one. -- Christopher Dobbs Marc Storck wrote: Will you have a Channel Bank for E1, many E1s instead of the FXS/FXO ports?? Marc Christopher Dobbs wrote: My company has started development on a Ethernet based channel bank. Here are the (current) spec's - 10/100 Ethernet Port - Up to 96 FXS/FXO ports (Thats 4 DS1's for the math impaired) - Serial Console - TDMoE - IAX2 - EETP (A protocol that we have designed for IP Telephony) We have just started prototyping this device, so... -- Christopher Dobbs ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ethernet Channel Bank (Comming Soon to a NOC Near You!)
My company has started development on a Ethernet based channel bank. Here are the (current) spec's - 10/100 Ethernet Port - Up to 96 FXS/FXO ports (Thats 4 DS1's for the math impaired) - Serial Console - TDMoE - IAX2 - EETP (A protocol that we have designed for IP Telephony) We have just started prototyping this device, so... -- Christopher Dobbs ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice Prompt Info
Your previous messages came through, but had "[Asterisk-Users] Re: Asterisk-Users Digest, Vol 5, Issue 158" as the subject. I for one usually skip messages where the person did not think to change the digest subject to something more meaningfully. To help others help you could those of you who get digest form please fix the subject before replying? Thank you in advance. -- Christopher Dobbs [EMAIL PROTECTED] wrote: I have sent this twice now but, I think, for some reason, it has been sent as HTML which is causing it to be drooped (and rightly so). I apologize in advance if, suddenly, those two make it though along with this one. Anyway, I should have been more clear in my original message. I am looking for departments that fit - into - those strings. Pretty much, if a person could replace DEPT with what they are thinking, they are on track. I mention the strings them selves only as a way to show context. When I first posted that message I had a handful of examples that did not fit into that 'mold' but, for the life of me, I can not think of one now. Thanks; James Date: Fri, 10 Dec 2004 16:24:00 -0800 You should not put the "press" or the number in the prompt. Have them as separate sounds, that way, they are more generic. [EMAIL PROTECTED] wrote: I am looking for titles that fit into the string: "press 1 for the DEPT department" or "press 1 for DEPT" but if you have other suggestions, let me know. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice Prompt Info
You should not put the "press" or the number in the prompt. Have them as separate sounds, that way, they are more generic. EG: Background(press-1-for) Background(sales) Background(and) Background(service) Background(department) Background(press-2-for) Background(Tech) Background(support) -- Christopher Dobbs [EMAIL PROTECTED] wrote: I am trying to put together a list of 'departments' to request as voice prompts. I have the biggies (sales, accounting, shipping, etc...) but I want to make sure I do not miss any. If anyone anyone has some suggestions (Ha... that is like going to an NRA meeting ans asking if anybody has a gun :-) ) please forward them to me (and / or post here although, with the volume of this list I do not always have time to read every digest so the 'and' option may be best.) so that I can compile a single list, verify that they are not already available, group them, and send them on. Please put 'voice prompt' in the subject line of anything you forward me so that I am less likely to miss it. I am looking for titles that fit into the string: "press 1 for the DEPT department" or "press 1 for DEPT" but if you have other suggestions, let me know. I will be collecting these for about a week so please try to get them to me in that time frame. I am hopeful that, with these prompts, it will be possible to make a complete (albeit fairly generic) tree, all with the same voice. Thanks; James alspachfam at charter dot net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ripping CD audio for MOH
Jean-Michel Hiver wrote: What applications (osx or linux) are best? Optimal settings? linux 'grip' is very nice. As is RipperX (http://ripperx.sourceforge.net/) -- Christopher Dobbs ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] four wildcards in a single pc
I have done multi-TDM4xx cards in a box. DO NOT TRY THIS!! Sorry for yelling but, Our system worked, and I use that term loosely, for about a week. The hardware simply cannot handle the onslaught of IRQ's that this causes. We got popping and beeping, and dead lines. We put each card in a separate machine, and used IAX to link them. All is working perfectly now. I am going to try TDMOE next. If you are interested in the results of that, contact me off list. (DO NOT just send me a message saying just "Yes I am interested" I have received several of those and the just get nuked 'cause I don't know what the are interested in) -- Christopher Dobbs Shoval Tomer wrote: Hi. Please excuse me asking this again. But it really puzzles me. We're installing asterisk at a branch office at NJ (HQ is at Petach-Tikva) It'll need to support 5 POTS lines, 11 analog extensions and four VOIP phones. I wanted to go with a T1 card from digium and a channel bank, but we have a dead line. It has to be up and running by January 1st. I don't have the time to start shopping at ebay, where you don't know what you'll get, and you need to install, under time pressure something you not familiar with. So I thought of installing a combination of four pci cards in the machine, and everybody on the list just keeps telling me it won't work. I have installed successfully more then four cards in a machine before. I had a firewall with eight network interfaces (one quad card, one duo and two singles) I have machines with two dialogic boards, a pci display card, and a network interface. And I know I've had machines at home that had a display adaptor, modem, network, scsi, and soundblaster all together. Yet, people claim it won't work because of lack of IRQs, and that it's not related to Digium. What am I missing? Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pc
I know that this has been replied to, but, where I work, We use the Carrier Access Group Adit 600 Channel banks. It supports two T1's and my boss tells me that we can get the for about 400-800 dollars. They use 8 port c\expansion cards, so you can three groups of ports, say 16 FXS (2x FXS) and 8 FXO ports (1x FXO). These units are programmed using a crossover cable and a serial port. They have a lot of other features that I don't know how to use, but they work grate for us. I hope this helps. -- Christopher Dobbs Shoval Tomer wrote: Can you recommend a channel bank make and model that will support all (or most) of Asterisk's features and can be installed by a newbie? I'll search for it on ebay -Original Message- From: Steven Critchfield [mailto:[EMAIL PROTECTED]] Sent: Wednesday, December 08, 2004 7:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] pc On Wed, 2004-12-08 at 19:09 +0200, Shoval Tomer wrote: -Original Message- From: Steven Critchfield [mailto:[EMAIL PROTECTED]] Sent: Wednesday, December 08, 2004 6:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] pc On Wed, 2004-12-08 at 18:07 +0200, Shoval Tomer wrote: I'm going to install asterisk with four digium cards. Can anyone mention a brand that carries boards with 4 compatible pci slots? What 4 cards are you thinking of installing? Most people seem to run into trouble after the second concurrent card. I was thinking one 4 port FXO card Two 4 port FXS card And the fourth with one FXO and three FXs For a total of 11 FXSs and 5 FXOs (for five POTS lines and 11 analog extensions). You would be far better off going with a channel bank and T1 card. Immediately you get room for expansion as you won't be limited to just 16 ports. As for cost, 4x $305(I think that is the current price) = $1220, $1220 - $500(t100P) = $720, or enough left over to get taken on parts from ebay a couple of times before you reach level costs. What trouble should I run into? Define trouble? Hard to configure, or impossible because of IRQ sharing issues or whatever? Trouble being 4 x 1000 interupts per second on the machine. Lack of expansion, interupt sharing, any number of annoyances. Anyways, almost all motherboards put 4 PCI slots on the board as it is not much extra and expected on anything non entry level. I thought so. Apparently most of them only come with three. I was hoping to locate a motherboard that has five so I'll have room to expand, but four will be great. Anybody knows if PCI express technology is compatible with Wildcards? -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Faxing..not 100%
I am trying to do exactly what you have done, can you send some examples to me. please reply to [EMAIL PROTECTED] so as I do not miss any reply you may send. -- Christopher Dobbs Andrew Kohlsmith wrote: On December 7, 2004 01:06 pm, Matthew Boehm wrote: POTS -> PRI -> Asterisk -> ATA (Fax) My setup: POTS -> PRI -> Asterisk -> Asterisk -> TDM430P (Fax) The asterisk-asterisk link is a dedicated 1024kbps SDSL link on dedicated ethernet cards (i.e. all other network traffic works is on the other network interfaces) Our faxing is pretty solid. Not *perfect* but certainly better than 50%. Anything unusual about your PRI? Do you hear crackles or "zapping" noises on it on voice calls? Is your * box overloaded? Anything funny with interrupts? Using a RedHat kernel? -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] gsm codec, very poor quality.
I use sox to do wav to gsm conversion and have not had a problem with sound quality. Another trick i use, i have a menu that all it is for is recording the menu prompts for the rest of the system. It was complicated to write, but, well worth it in terms of creating the menu prompts I need. Jon Radon wrote: Sorry this doesn't answer your question. Any reason to not leave them as wav's? On Tue, 7 Dec 2004 10:42:58 +0100, Matthew Oulton <[EMAIL PROTECTED]> wrote: Currently I am creating .wav files and then converting them via SOX to .au file format, then running them through a gsm codec convertor which all works fine except that it sounds like the recording was made with a sock in my mouth !! Could someone in * land help me to get a good sound quality with gsm format. Thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Kind of off-topic: VoIP services and multiple callers
I disagree completely, We provide VOIP services (in the final testing phases), and there is more interest in unlimited than in per minute rates. We have run the stats and our unlimited plans are prices so that we get more than what the end user uses. (on average, that is) -- Christopher Dobbs nik martin wrote: Andrew Kohlsmith wrote: On December 6, 2004 10:12 pm, Michael Giagnocavo wrote: Except the providers who offer "unlimited" -- in that case, they want you to use as little as possible, so they can make their money. They're the ones that are on the way to bankruptcy. EXACTLY ;) Aint no free lunch, my friends. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with music over intercom.
I am using Console/DSP for an intercom. I want to play my MP3 collection over it when no one is using it, like when they do in the supermarket. Can anyone help me with this. Any suggestions will be appreciated. -- Christopher Dobbs ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Changing Asterisk Voicemail Storage Location
If a patch is developed that will acomplish this division, I am interested in it. My company is planning on deplying a massive * network with a central server providing VM. This would make the VM server easyer to admin. -- Christopher Dobbs Adam Goryachev wrote: On Thu, 2004-11-25 at 16:22, Java Rockx wrote: Can anyone tell me how difficult it would be to change the way asterisk stores/retrieves user messages as follows? Currently mailboxes are in /var/spool/asterisk/voicemail/{context} But I need to store messages in a hash to limit the number of directories per context. All mailbox extensions are the user's 10-digit phone number (aka, DID). The parts of a DID are as follows So my hashing would look like this /var/spool/asterisk/voicemail/{context}/{npa}/{nxx}/{line} And in the {line} directory we would have the usual Asterisk files/directories for inbox, etc. We're looking at a large number of mailboxes and this would give us a maximum of 1 mailboxes per directory - which plays nice with the Linux file system. You might look at alternative filesystem formats. "Linux file system" is not any file system I've heard of. Most likely you are referring to the filesystem that you get by default when you do an install and just click next without understanding the option each step of the way. Specifically, look at reiserfs, it is very good at handling directories with large number of files, as frequantly seen in mail servers using maildir format etc... I'm not sure I understand all the details, but reiserfs should be equivalent in speed to a DB at least, I've frequantly seen it referred to in that way back when I used to subscribe to their mailing list. I suppose you might ask the question, is it faster to parse the mailbox name in userspace and then look up the correct file, or let the kernel parse the name, and find the file for you Hope this helps you... Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fw: Gift for Mark Spencer
You, bloody moron. Is not most email unsolicited. I never asked you to send an email, Your message is off topic, Your getting rude, therefor YOU are a spammer. Now, in the name of all that is decent, drop this thread. Lets get back to what we are all here for, discussing Asterisk. -- Christopher Dobbs Systems Manager Eracew Computer Services Joe Greco wrote: On Tue, 2004-11-23 at 19:06 -0600, Joe Greco wrote: [..snip..] On the flip side, senders of spam should not expect recipients to go to much (or any) trouble on their behalf, especially given the current spam environment on the 'net. They - not hackerwaCker - blew the surprise by sending the message to recipients unknown. I'll just summarise all you said into one conclusion which remains the same as to what Steven said: hackerwanker is a moron. :-) No, that's not what I said. If you want the short, brutal summary, it'd be: The spammer who sent the message is the moron. Really, there are all sorts of bizarre phishing schemes and other scams out on the 'net. If you go asking random people for donations, and cannot put the request in the context of solid well-knowns, such as an organization or individual who is clearly legitimate, then it looks quite possibly like a scam of some sort, and posting it to the list isn't exactly unreasonable - it's more like a "watch out for this scam" community service. However, we also have to remember that even being a well-known wouldn't make it right to send unsolicited bulk e-mail. So. It's unfortunate (for the people trying to organize the gift) that hackerwacker sent an alert to the list. It's not unusual, though. As service providers, many of us actively encourage customers to put a stop to abuses of the mail system such as chain letters and other scams by asking people to take active countermeasures. I'd consider this to be an example of just such a countermeasure. It seems fitting that spammers should not have their goals furthered by the act of spamming. It would seem that this is precisely what happened in this case. I'll further note that I did receive a copy of the spam in question. While I did not choose to complain to the relevant sites about it, or to post a message to the mailing list, the Boulder Pledge is certainly applicable - I will not be contributing towards a gift effort that spammed. ... JG ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Secure IAX Communications
I have been moitoring your discutions about Secure IAX. I have been thinking about writing a wrapper for IAX that uses OppenSSL to encrypt the IAX stream. If enough prople are interested in this I will do it. If you are interested, email me offlist at [EMAIL PROTECTED]. if I do this I will be submitting it to Digium for inclusion in mainstream Asterisk. But I will only spend the time to do this if there is enough interest. (While I am at it, If some one knows anything about ADSI, I would like to add support for ADSI to IAX.) Christopher Dobbs Software Engineer Eracew Computer Services http://www.eracew.net/ [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The Apperiant Death of IAXtel
I did not know what enum was, or where it was. I just knew that IAXtel has been down most of the time that I have tried them. Thank you for the update, but my offer still stands open. If any one is interested, just email [EMAIL PROTECTED] I in no way am trying to compeate with, or replace ENUm or IAXtel, just trying to be a resource. Duane wrote: Christopher Dobbs wrote: What ever, Just trying to be a help to the system. The benefit of enum over a relay service is the ability to interoperate with others using SER and other VoIP PABXs as well, rather then being limited to just other asterisk users, it's self managed via the web interface so people can keep their numbers and hostnames etc up to date themselves... /2cents... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The Apperiant Death of IAXtel
What ever, Just trying to be a help to the system. Duane wrote: Christopher Dobbs wrote: Let me clerify, Send a username and password for use on my new IAX relay system. It wont use real phone numbers, but it will work to link the /free world/ of IAX users. Why not just use www.e164.org via enum lookups then, it does let you use both real phone numbers (after verification) and non-real numbers... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The Apperiant Death of IAXtel
Send account request information to [EMAIL PROTECTED] Christopher Dobbs wrote: Let me clerify, Send a username and password for use on my new IAX relay system. It wont use real phone numbers, but it will work to link the /free world/ of IAX users. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The Apperiant Death of IAXtel
Let me clerify, Send a username and password for use on my new IAX relay system. It wont use real phone numbers, but it will work to link the /free world/ of IAX users. Kevin Walsh wrote: Christopher Dobbs [EMAIL PROTECTED] wrote: Given that IAXtel has not been responding for some time, I am willing to setup accounts for thoes who want to have that kind of functionallity. If you are interested, send me an email with your requested username and password, and i will send you your account information. Given that IAXtel has not been responding for some time, what use is a username and password? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] The Apperiant Death of IAXtel
Given that IAXtel has not been responding for some time, I am willing to setup accounts for thoes who want to have that kind of functionallity. If you are interested, send me an email with your requested username and password, and i will send you your account information. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX and ADSI Help
Seth Remington wrote: On Mon, 2004-11-08 at 17:47, Christopher Dobbs wrote: Does anyone know how to transfer ADSI information over IAX, I have looked at the code, and it apears that this is posible. I think ADSI currently only works with Zap channels. You are correct that it should be possible with any channel type but my understanding is that it's only currently implemented in the Zap channel driver. I don't have paperwork to back that up though :) What are you trying to do? Run an analog ADSI phone through an IAXy or something? -Seth I am trying to extend my phone service to my dads house, we both are connected to the same wireless internet provider. I want to be able to access my voice mail while at his house, and I want to be able to use the ADSI VM interface. Further more I am writeing an ADSI program for configuring the call routing of the asterisk PBX. I have 4 phone lines and I want to export 2 of them to my fathers house. (Well actuly They are already there, but not using ADSI) The IAX source has information it it about ADSI, but I dont understand it well enough to know how to transfer the ADSI stream. (If fact, it apears that IAX maintains a flag as to wether or not the other end supports ADSI) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX and ADSI Help
Does anyone know how to transfer ADSI information over IAX, I have looked at the code, and it apears that this is posible. -- Christopher Dobbs Software Engineer Eracew Computer Services ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users