Re: [Asterisk-Users] AVM ISDN Fritz!Card USB works
On Tue, Dec 16, 2003 at 11:28:38AM +1100, Gonzalo Servat wrote: > On Tue, 2003-12-16 at 10:34, Michiel Betel wrote: > > Is case anyone wants to know... The Fritz! USB ISDN box works fine with > > Asterisk! > > I'm running CAPI 0.3.0 and love it, because the mini ITX server I have > > only takes one PCI slot which is now filled with a 4 port Digium card. Is this the micro PABX model 4FXS + 1 ISDN FXO (USB 3.0) or the plain 1 ISDN FXO (USB 2.0)? cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax
On Thu, Dec 04, 2003 at 10:50:57AM +1030, [EMAIL PROTECTED] wrote: > Hi > > I have a second line that we use for a fax server > > Since we are luck to get 2 faxes a week > > I want to use this line as a dial out line for * > > But still need to be able to send and receive faxes on it > > Has anyone got any ideas how I could accomplish this ?? Here's a strategy off the top of my head: 1. plug the line into your voicetronix & the fax machine 2. configure asterisk to use it as the main dial-out line 3. configure asterisk to not pick up calls incoming on that line 4. configure users to check that the fax machine wasn't in use before dialing out, or to dial a special dial-out code to use the other line, so they don't stuff your faxes. cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN Cards available in Australia
On Tue, Nov 25, 2003 at 10:30:52AM -0500, Kimble Young wrote: > Hi, > > I'm after some ISDN card resellers in Australia. I've noticed the AVM cards > should work well here, however I've found the reseller here has no website, > is listed in Yellow pages as management consultants (??) and when I called > quoted almost $400. Well he got 2 cards to me overnight 2 weeks ago. Also he gambled and put up the cash to get the cards approved, and the Australian approval system says you have a choice of paying a premium to him (~$200/card) while he recoups his costs, or getting your own approval (~$15,000), or risking a fine (~$17,000), or getting written permission from your Telco (free, has anyone tried?). > Does anyone know where I can get the AVMs in Australia a bit cheaper or > another card (not the Netjet) that works with Asterisk. The other A-ticked cards start with OpenLine4 (~$1,000) for 4 port PSTN FXO, or you can but Eicon Diva cards for ~$1,200 for 1 port BRI or ~$3,100 for 4 port BRI. If you don't care about the A-tick then you can import your own AVM Fritz. Hope this helps, cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fritz pci / chan_capi / australia setup
These instructions are mostly good, I have some corrections below...
On Thu, Nov 20, 2003 at 09:14:24AM +1100, Anthony Wood wrote:
> Hi * Fans,
>
> I have some fritz cards now, followed instructions from stuart hirsts email of Jun
> 28:
>
> ->
> Thanks for your info but I think I have it working at last. Below are
> the steps I took which might help others.
>
> 1) Download the PCI AVM drivers from ftp://ftp.avm.de/cardware
> 2) Download the Chan_capi from http://www.junghanns.net/asterisk/
> 3) tar -xvzf fcpci-suse8.0-03.09.10.tar.gz which creates a directory
> called "fritz"
> 4) edit the file fritz/src.drv/tools.c and put "//" in front of the
> following lines to exclude them :
>
> #include
> #include
> #include
> #include
> //#if LINUX_VERSION_CODE < KERNEL_VERSION(2,4,18)
> #include
> //#else
> //#include
> //#endif
>
> This stops the make error message.
>
> 5) run "make"
> 6) run "make install"
> 7) create a file /etc/capi.conf with the following line in it:
> fcpci - - - - - -
7.5) You should stop any isdnutils services and unload any hisax or isdn modules at
this point
> 8) run "insmod -f fcpci". You will get errors about the version of GCC
> but it should still work
> 9) run "capiinit"
> 10) run "capiinfo" which should give output showing the card and
> supported features.
> 11) extract using tar -xvzf chan_capi.0.2.2.tar.gz
> 12) in the chan_capi-0.2.2 directory run "make" and "make install" and
> "make config"
> 13) edit /etc/asterisk/modules.conf and add :
>
> load => chan_capi.so
>
> And in the [global] section add:
>
> chan_capi.so=yes
>
> 14) then start Asterisk
>
This next bit is apparently unnecessary/no good.
> Also forgot to mention that you should make sure that the "isdn" and
> "hisax" modules are loaded by doing :
>
> modprobe isdn
> modprobe hisax
>
> I now have incoming calls working OK and working on getting outgoing
> working.
>
> Stuart
>
> <-
>
> OK, so I have some good signs:
>
> forge:/etc/asterisk# cat /proc/capi/controllers/1
> name fritz-pci
> io 0xD400
> irq 10
> type A1
> class14
> ver_driver 3.09-10
> ver_cardtype fritz-pci
> protocol DSS1
> linetype point to multipoint
> forge:/etc/asterisk#
>
> *CLI> capi info
> Contr1: 2 B channels total, 2 B channels free.
> *CLI>
>
> I've set up capi.conf as best I can:
> forge:/etc/asterisk# grep -v ^\; capi.conf | grep -v ^$
> [general]
> nationalprefix=0
> internationalprefix=0011
> rxgain=0.8
> txgain=0.8
> [interfaces]
You should list the msns that your devices connect to here:
> msn=292996337
> incomingmsn=*
> controller=1
> softdtmf=1
> accountcode=
> context=demo
> devices=2
> forge:/etc/asterisk#
>
> And some extensions:
> grep -i 6337 extensions.conf
> exten => 11031976,1,Dial,CAPI/292996337:0416059875,5,tr
> exten => 02111947,1,Dial,CAPI/@292996337:0416059875,5,tr
>
> And this is what I get every time:
>
> *CLI> NOTICE[81926]: File chan_sip.c, Line 5018 (handle_request): Failed to auth
> enticate user "woody" ;tag=f7f432a8-0b96-4daa-901a-e21e153cf
> f96
> -- Executing Dial("SIP/woody-8607", "CAPI/@292996337:0416059875") in new stack
> -- data = @292996337:0416059875
> -- capi request omsn = @292996337
> == found capi with omsn = 292996337
> == CAPI Call CAPI[contr1/292996337]/1 -- Called @292996337:0416059875
> -- CONNECT_CONF ID=001 #0x0005 LEN=0014
> Controller/PLCI/NCCI= 0x101
> Info= 0x0
>
> == received CONNECT_CONF PLCI = 0x101 INFO = 0
> == DISCONNECT_IND PLCI=0x101 REASON=0x3301
> -- CAPI Hangingup
> == No one is available to answer at this time
>
> REASON 0x3301 means Layer 1 problem or signalling killed the channel.
The cable was good, but the NT unit I have (Telstra Australia NT1 plus II) disables
the S-BUS
when you have a USB cable in it (undocumented feature).
cheers,
--
Woody
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[Asterisk-Users] fritz pci / chan_capi / australia setup
Hi * Fans,
I have some fritz cards now, followed instructions from stuart hirsts email of Jun 28:
->
Thanks for your info but I think I have it working at last. Below are
the steps I took which might help others.
1) Download the PCI AVM drivers from ftp://ftp.avm.de/cardware
2) Download the Chan_capi from http://www.junghanns.net/asterisk/
3) tar -xvzf fcpci-suse8.0-03.09.10.tar.gz which creates a directory
called "fritz"
4) edit the file fritz/src.drv/tools.c and put "//" in front of the
following lines to exclude them :
#include
#include
#include
#include
//#if LINUX_VERSION_CODE < KERNEL_VERSION(2,4,18)
#include
//#else
//#include
//#endif
This stops the make error message.
5) run "make"
6) run "make install"
7) create a file /etc/capi.conf with the following line in it:
fcpci - - - - - -
8) run "insmod -f fcpci". You will get errors about the version of GCC
but it should still work
9) run "capiinit"
10) run "capiinfo" which should give output showing the card and
supported features.
11) extract using tar -xvzf chan_capi.0.2.2.tar.gz
12) in the chan_capi-0.2.2 directory run "make" and "make install" and
"make config"
13) edit /etc/asterisk/modules.conf and add :
load => chan_capi.so
And in the [global] section add:
chan_capi.so=yes
14) then start Asterisk
Also forgot to mention that you should make sure that the "isdn" and
"hisax" modules are loaded by doing :
modprobe isdn
modprobe hisax
I now have incoming calls working OK and working on getting outgoing
working.
Stuart
<-
OK, so I have some good signs:
forge:/etc/asterisk# cat /proc/capi/controllers/1
name fritz-pci
io 0xD400
irq 10
type A1
class14
ver_driver 3.09-10
ver_cardtype fritz-pci
protocol DSS1
linetype point to multipoint
forge:/etc/asterisk#
*CLI> capi info
Contr1: 2 B channels total, 2 B channels free.
*CLI>
I've set up capi.conf as best I can:
forge:/etc/asterisk# grep -v ^\; capi.conf | grep -v ^$
[general]
nationalprefix=0
internationalprefix=0011
rxgain=0.8
txgain=0.8
[interfaces]
msn=292996337
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=demo
devices=2
forge:/etc/asterisk#
And some extensions:
grep -i 6337 extensions.conf
exten => 11031976,1,Dial,CAPI/292996337:0416059875,5,tr
exten => 02111947,1,Dial,CAPI/@292996337:0416059875,5,tr
And this is what I get every time:
*CLI> NOTICE[81926]: File chan_sip.c, Line 5018 (handle_request): Failed to auth
enticate user "woody" ;tag=f7f432a8-0b96-4daa-901a-e21e153cf
f96
-- Executing Dial("SIP/woody-8607", "CAPI/@292996337:0416059875") in new stack
-- data = @292996337:0416059875
-- capi request omsn = @292996337
== found capi with omsn = 292996337
== CAPI Call CAPI[contr1/292996337]/1 -- Called @292996337:0416059875
-- CONNECT_CONF ID=001 #0x0005 LEN=0014
Controller/PLCI/NCCI= 0x101
Info= 0x0
== received CONNECT_CONF PLCI = 0x101 INFO = 0
== DISCONNECT_IND PLCI=0x101 REASON=0x3301
-- CAPI Hangingup
== No one is available to answer at this time
REASON 0x3301 means Layer 1 problem or signalling killed the channel.
Is there something I have missed?
thanks in Advance
--
Woody
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Re: [Asterisk-Users] creative VoIP blaster & *
On Wed, Nov 19, 2003 at 06:07:46AM -0500, Patrick Cantwell wrote: > Ok, > I've googled for 15+ minutes, and have yet to find a usable answer, so I'm > going to annoy everyone and ask here. > > I have, in my posession, a creative VoIP blaster. I have installed the > fobbit LKM and I can see the device. Can I use it with asterisk in any > meaningful way, shape, or form? I'd love to be able to buy an IP phone, > ATA, or FXO card, but lack the funds at the moment (won't get into why a > grandstream phone at $65 is out of my budget, just take my word for it). > > Can I turn this hardware that's laying around into anything useful? I saw on an Asterisk FAQ somewhere (I too can't find it in Google) but it's here: http://www.asstricks.org/faq.html -> 4. I bought a Creative Labs VOIP-Blaster. How can I hook it up to use with Asterisk? You may hook up the VOIP-Blaster to Asterisk, but you can only use it to talk to other VOIP-Blasters. Due to the patent on the codec used by the VOIP-Blaster, it isn't likely that Asterisk will allow the VOIP-Blaster as a simple handset (other than with other VOIP-Blasters) anytime soon. -> Anyway, maybe you can find another blaster user, or ask Creative what if you can implement the codec :-) cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anybody using Sphinx
On Wed, Nov 19, 2003 at 10:22:55AM +0800, Steve Underwood wrote: > Arnold Ligtvoet wrote: > > >Since I would like the user names to be auto-generated by the system, I > >would guess that this could best be done using festival with a localized > >voice. I think there is a Dutch voice for Mbrola with should integrate into > >festival ( note to self : need bigger harddisk :-) ) > > > > > Speech recognition accuracy is not great under ideal conditions. Doing > what you suggest seems unlikely to achieve any meaningful accuracy. > Speech recognition training systems require many occurances of a word or > phrase, clearly spoken, before their accuracy becomes useful. A one shot > utterance from Festival seems to fail on both counts :-) > Sphinx isn't doing general speech recognition, it is determining which of a list of patterns it has you said, like mobile phones do. So it's fairly easy to tell between "Jennifer" and "Frank" if there are no other options. When you call directory assistance in Australia, the IVR asks you what name you want, and gives you a suggestion out of the top 100 or 200 names, which you can accept or reject. Makes for riducule, but beats waiting on hold. > Bottom line: the very best speech recognition still sucks. As a British > speaker I never get more than about 40% accuracy speaking into a US > trained recogniser. I have never had better than about 70-80% accuracy > on a British trained recogniser. Strangely, my terrible Cantonese gets > nearly 100% on SpeechWorks recogniser. :-\ This is true for general speech recognition, where the computer has a much larger dictionary to match the sound waves against. cheers -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] This is how you Search the Archives
Go to www.google.com type in your search query add this to the end of your search query: site:lists.digium.com e.g. http://www.google.com.au/search?hl=en&ie=utf-8&oe=utf-8&q=Australia+site:lists.digium.com The mailing list used to be on www.marko.net, I'm not sure if the whole archive was moved across, you might want to search with site:www.marko.net OR site:lists.digium.com cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO Card/Interface for Australia
On Wed, Nov 19, 2003 at 10:10:13AM +1100, Alexander Romanov wrote: > Hi Guys, > > Pardon me for the might be stupid/repeated question ( I am new to the > game), but has anyone successfully implemented any FXO cards in > Australia. As far as I know the current A-ticked choices are: AVM Fritz!Card (from AU distributor) - works well NetJet - works well but echo issues for internal callers Voicetronix Openline4 - works, but I don't know how well You need an A-tick on your device or written permission from your Telco before you can plug anything into the telco's network. If you import your own Fritz, it doesn't have an A-tick, so it is not legal despite being identical. > Particulalry I am thinking about X100P card. Is it good for local lines? If you want to risk the law, or get permission from your Telco, they work as well here as in the US, I've heard. Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO Cards in Australia
On Mon, Nov 17, 2003 at 04:32:50PM +1100, Gonzalo Servat wrote: > On Mon, 2003-11-17 at 16:00, Anthony Wood wrote: > > ISDN (telstra Onramp 2) is very similar in price to standard telstra lines. > > The only problem is you can't have ADSL & ISDN on the same line. > > > > We upgraded from 2 analogue lines to 2 digital (i.e. 4 channels) for $250. > > I was a bit turned off by the $300+ installation cost. I just rang > Telstra and its infact $190 if you already have a telephone line, which > I do. Awesome! > > How come you 4 channels if you only have 2 digital lines? I thought it > was one channel per line. I was told by the Telstra rep that I need a > OnRamp2 which is 2 channels, 2 lines. Yeah OnRamp2 replaces 1 analogue line, so we converted 2 analogue lines to 2 * OnRamp2 i.e. 4 lines. > > But they Telstra'd up the installation so we asked for (and got) the $250 waived. > > Typical (about Telstra'ing the installation, not the setup fee > discount!) > > > It's worth thinking about it because of the Advantages of Digital signalling when > > using voice: > > > > Know which number was dialed > > Know callerid early > > Know when the other end has hung up > > Better voice quality > > > > Using Analogue with Asterisk seems to be filled with Kludges to detect hangups, > > busy, etc. With ISDN, the exchange does that for you. > > Yeah, we're now looking at it again. Local calls are pretty cheap too as > long as you don't talk for too long. > > You mentioned echo problems with the NetJet cards. Is this still the > case or was it last time you tried that it that had echo problems? I did > a Google search and didn't find much on the echo problems with them. There is still the problem, so bad that 4 person business I know stumped up the cash for an ISDN10 PRI install (AU$2000) and a TE410P (AU$3000) to replace a netjet ($250). I have only heard good things about the AVM Fritz!Cards with chan_capi. They are more expensive than the NetJets, but cheaper per line than the Openline4. > Thanks again for the good info. I prefer Vanilla Coke to beer. :-) cheers -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO Cards in Australia
On Mon, Nov 17, 2003 at 03:49:40PM +1100, Gonzalo Servat wrote: > On Mon, 2003-11-17 at 12:20, Anthony Wood wrote: > > > I have spoken to a number of Australian users who are successfully using: > > > > X100P > > NetJet (echo issues) > > AVM Fritz!Card > > > > I hope to add myself to their number shortly, since we have recieved our Fritz!es > > > > Also [EMAIL PROTECTED] seems to be having some success with the VoiceTronix > > openline4. > > > > All these cards are legal except the X100P. > > Thanks very much Anthony. VoiceTronix cards are a little out of my > budget, the NatJet & AVM cards are for ISDN (and we need standard > analogue). ISDN (telstra Onramp 2) is very similar in price to standard telstra lines. The only problem is you can't have ADSL & ISDN on the same line. We upgraded from 2 analogue lines to 2 digital (i.e. 4 channels) for $250. But they Telstra'd up the installation so we asked for (and got) the $250 waived. It's worth thinking about it because of the Advantages of Digital signalling when using voice: Know which number was dialed Know callerid early Know when the other end has hung up Better voice quality Using Analogue with Asterisk seems to be filled with Kludges to detect hangups, busy, etc. With ISDN, the exchange does that for you. cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO Cards in Australia
On Mon, Nov 17, 2003 at 12:13:09PM +1100, Gonzalo Servat wrote: > Hi All, > > This topic has come up before in the Asterisk mailing list many times, > so I know that a lot of people have given up in waiting for a FXO card > to be approved by the Australian telecommunications authority. My > question is: all legalities aside - is anyone using a FXO card in > Australia successfully? I have spoken to a number of Australian users who are successfully using: X100P NetJet (echo issues) AVM Fritz!Card I hope to add myself to their number shortly, since we have recieved our Fritz!es Also [EMAIL PROTECTED] seems to be having some success with the VoiceTronix openline4. All these cards are legal except the X100P. cheers, Woody PS: You are a SLUG member, no? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Couple of Questions for Australian Users!
On Fri, Nov 14, 2003 at 03:41:10PM +1100, John Paine wrote: > > > -Original Message- > From: Adam Hart [SMTP:[EMAIL PROTECTED] > Sent: Friday, November 14, 2003 9:47 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Couple of Questions for Australian Users! > > > Just a couple of questions for Aussie users/resellers! > > > > I have only just started to look at asterisk a couple of weeks ago and > > have found some very intresting discussions and some useful info on what > > can and can?t be done with it and technology. > > > > The questions i have is, > > > > Are there people using it is Aussie?( I would say yes so prob answered > > my own question ) :) > > There's quite a few, not all use this list (yet no one wants to test my > australian ringtone patch) I've come across about 10 I think, including myself, but I assume there are more. > > Hardware, is the digium hardware OK in australia? or does it require > > changes/mod?s or is just a pain in the butt? > > > It works but most of it isn't A-tick'd, therefore it's illegal to use ($20k > fine). The TE410P is the exception. Depends what connecting to, search the > mailing list archives for specifics. > > > Alternate hardward for Aussie if above is an issues? > > > There's alternatives but that discussion isn't received well in the mailing > list. Most people i believe just use digium hardware regardless of A-tick. The AVM Fritz!Cards are the A-ticked small installation alternative. The plug into the Telstra ISDN Onramp2 which isn't too expensive to get (They had a sale on recently, upgrade your POTS line to ISDN for AU$125. The Telstra'd it up so we asked them to waive the installation and they did). The way the A-tick system works is that you have have the sticker on the card, so you have to buy it from whoever payed the $$$ to get it A-ticked, so they can recoup their costs. So you have to get them from the Australian Distributor who will express post them to you, we got ours on Wednesday. You can also use NetJet ISDN cards, but they have echo issues on the internal side. If it has to be POTS, Voicetronix has an OpenLine4 card which works with Asterisk, I don't know what the quality is like. Voicetronix is also getting the OpenSwitch 12 card certified, it can work as FXS or FXO or a mixture (0/4/6/8/12 ports FXS) > > Resellers/Importers of the Digium hardware? I have looked at the website > > and see that there are aussie resellers but they all seem to have their > > own agenda and the one I spoke to was not intrested in selling me > > hardware alone they wanted to do me a whole service deal! > > Buy straight from digium :) If digium puts the A-tick on it, you'll be right. I don't think the TE410P is finished the paperwork yet, I assume digium will post you the A-tick sticker later if you buy one now :-) > > Also IP phones? What type/recomendation? I am going to be using Phone > > Software as well on laptops but I need an IP phone option to do phones > > over WAN. I really only know of the cisco and avaya types, I have not > > had any contact with any other types of phones. Intrested in Aussie > > suppliers for what ever phone you would recomened unless they are great > > phones then I would have to find an USA reseller. > > > I don't know any aussie resellers Nor I > > OK think thats all! Thanks Guys doing some great work with what looks to > > be some very GREAT software! I can't talk all the credit :-) > > Thanks in advance for your help! Hope it is help. -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] D-Link modem with *
On Fri, Nov 07, 2003 at 02:12:07PM +0530, ranga wrote: > Can we use D-Link external modem with asterisk? You are a bit vague on the details, so I will be vague on my answer: maybe not. Modems with linux don't generally do full duplex voice. Most users use Asterisk with Digium Hardware or other specialized voice hardware. cheers, Woody PS: What do you want to use it for? (Data/Fax/Answering Machine/Voice/Doorstop)? PPS: What type of D-Link external modem is it and what connections does it have? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterix - Digium replacing our current key system
On Thu, Nov 06, 2003 at 12:42:19PM +1100, Drew Foehn wrote: > Hello, > > I'm posting to the list to get some ideas on whether or not asterisk and digium > products would be right for my organization. We have 3 lines that comes into Since you are in Australia, I assume these are 3 plain lines from Telstra. > a LG digital key (Aria-43e) system, with Digital LG phones. I have two > questions; could we replace the digital key system with a PC running Asterix > and keep the same phones? Also We have about 14 users and 2 offsite offices Depends on what standards the phones use. At worst you can put Asterisk between your Aria and your phone lines to keep using the Aria Handsets. At best you can just use the phones with an ADSI channel bank. If the phones have any Austel stickers on them then likely you can. > and we are looking at saving costs from our long distance by using VoIP and our > ADSL line, what would be the ideal setup for this? I think the best way forward is to convert 2 of your lines to ISDN (giving 4 voice lines), leave the 3rd one as ADSL/fax, and connect to asterisk with 2 AVM Fritz!Card PCI from the Australian distributor. To connect the phones, either get a Yank Channel bank and a T100P if the phones work that way, or plug your Aria into a T100P or E100P if it has those interfaces, otherwise into a TDM400P or two, depending on how many lines it supports. The X100P is not A-ticked, neither are most yank channel banks, so you need written permission from the Telco to go down that route. cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The Minimum Cost of Setting up an Asterisk Phone System?
On Wed, Nov 05, 2003 at 02:03:55PM -0700, Steve Murphy wrote: > Everyone-- > > Here's a cost analysis, rather crude and inspecific, of using Asterisk > to implement a phone system. I'm really quite naive and new to all this, > so I'd appreciate any corrections, tips, pointers, etc, from those in > the community, who have far greater experience, knowledge, skill, etc. > than I. Am I forgetting something important? Am I way off in my > estimates? > > > The Minimum Cost of setting up an Asterisk Phone system: > > Fundamental Building Blocks: > > 1. No more phones serviced than one computer can handle. > 2. Computer = self-built or whatever, approx. $500 > 3. OS = Linux. $0 > 4. Phones. >Cheap Touch-tone phones: $30 each (Estimate at what I can get > at Walmart, quantity one purchases. No digital readout, no > programmable features. >Voip Phones $250 estimated per-phone average cost. Realize that > costs can vary widely here! $65 for Grandstream? > 5. Channel Banks. Looking at what's available on Ebay, I estimate you >should be able to pick up a fully loaded, 24-channel FXS/FXO bank for >$650 average. Low = $200 High = $4500. I have no way of telling >which channel banks are compatible with asterisk. Assume that they >are. > 6. Digium cards: FXO card = about $100. FXS card = $125. > 4 port FXS=$300. Prices approx. See their web site >for exact prices. > quad span T1 (96 lines total) $1500 > Single span T1 (24 lines) $500 > 7. Wiring. Cost of Wiring is not calculated. Assume that the premises >is wired, with patch panels, closets, etc., already in place. > 8. UPS, power supplies, etc: Not specifically included in the estimates. > > 9. Used equipment can be cheaper, but: if you need a lot, you may not > necessarily be able to wait around for everything you need to show up on > ebay. And what you get may not be what you wanted, etc. 10. Time 11. Expertise (will save you heaps of time) > > Scenarios: > > 2x4 system: > Computer: $500 > 2 FXO : $200 > 1 4-FXS : $300 > 4 phones: $120 > -- > Total: : $1120 cost/phone = $280 > > 2x8 system: > Computer: $500 > 2 FXO : $200 > 2 4-FXS : $600 > 8 phones: $240 > -- > Total: : $1540 cost/phone = $192.25 > > 96 line system ( FXS/FXO mix 88/8) > Computer : $ 500 > QspanT1 : $1500 > 4 ChanBks: $2600 > 88 phones: $2640 > > Total: : $7240 cost/phone = $82.27 > > 192 line system (FXS/FXO mix 176/16) > Computer : $ 500 > 2 QspT1 : $3000 > 8 ChanBks: $5200 > 176phones: $5280 > > Total: : $13980 cost/phone = $79.43 > > Voip 2 x 4 System using gnophone on desk computers, already networked: > Computer : $ 500 > 2 FXO: $ 200 > 4 gnophones: $ 0 > > Total: : $700 cost/phone = $175 > > Voip 2 x 4 System using VOIP phone: > Computer : $ 500 > 2 FXO: $ 200 > 4 VOIPs : $1000 > > Total: : $1700 cost/phone = $425 > > Voip 2 x 8 System using gnophones on existing systems on existing > network: > Computer : $ 500 > 2 FXO: $ 200 > 8 gnophones: $ 0 > > Total: : $700 cost/phone = $87.50 > > Voip 2 x 8 System using Voip phones: > Computer : $ 500 > 2 FXO: $ 200 > 8 VOIPs : $2000 > > Total: : $2700 cost/phone = $337.50 > > > Voip 24 x 192 Phones, using gnophone on existing comps & network: > Computer : $ 500 > 1spanT1 : $ 500 > 1 ChanBnk: $ 650 > 192 gnoph: $ 0 > > Total: : $1650 cost/phone = $10.18 Voip 24 x 192 phones, using Grandstream Voip Phones: Computer : $ 500 1spanT1 : $ 500 1 ChanBnk: $ 650 192 GS100: $13440 switch/hubs: $ ? Total: : $15090+ cost/phone = $78.59+ > > > The fantastically low prices of using gnophones on existing systems is > only available IF: > 1. Your desktops are all running linux > 2. Your sound cards are compatible with linux/gnophone >(for me, the sound cards/systems seem to be the > stumblingblock/hurdle to get past!) Maybe X-ten X-lite or X-Pro for windows is good too. > Cheap $30 phones are an option, but you will not get: > Intercom
Re: [Asterisk-Users] isdn, modem, etc.
On Tue, Nov 04, 2003 at 02:43:36AM +0300, Shoval Tomer wrote: > I see many posts about using ISDN cards and modem cards. In my list lurking, I have gathered: ISDN comes in two flavours, Euro and US, and two sizes BRI(2 lines)/PRI (24-30 lines) hardware generally only supports one size and flavour. ISDN has many advantages because the signalling (e.g. The person you are ringing picked up, hung up, callerID etc.) is digital already and is well defined. If you are using analogue lines all this info is learned through what seem to me to be hacks. for more than 10-14 lines it is recommended to go with ISDN PRI (this is probably called T1 or E1 in your country) for 1 or 2 lines use a digium X100P per line, but there are some issues with UK CallerID, Asterisk not hanging up, and hardware not approved in certain countries for in between you can use ISDN BRI - either an ISDN4linux supported device (e.g. NetJet) or a CAPI supported device (e.g. AVM Fritz!Card PCI) I only hear good things about the CAPI devices and drivers (chan_capi). The NetJets give echo to your internal phone users using chan_modem_i4l, but sound ok to those on the outside. > Does this mean I can use a regular modem (that supports voice) as an FXS, or > FXO ? No. Most modems only support half-duplex voice over serial, and the linux stuff only supports half-duplex voice AFAIK. Maybe you can hack something together, I haven't heard of anyone using it in any environment yet. > If so, can you provide an example. there are drivers for chan_modem_aopen and chan_modem_bestdata, but I don't know if they work or not. > It might be the easiest way to test Asterisk's collaboration with the PSTN (or > analog phones) and provide proof of concept to non-believers. No, this is not the easiest way. I think the easiest way is to order a Dev Kit from digium, or if you have an ISDN BRI line handy, an AVM Fritz!Card Dev Kit from junghanns.net (the chan_capi developer) together with a TDM20P. (2 FXS). If you are keen and have more time than money, many people would like you to program your own channel (a combination of chan_local and chan_modem which uses a modem and a soundcard to do FXO full duplex. FXS requires something to generate the ring voltage (~75 Volts), which modems don't have. cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
On Mon, Nov 03, 2003 at 09:46:40PM -0400, William Waites wrote: > On Mon, 3 Nov 2003 17:10:10 -0600 (CST), Martin Pycko wrote > > It doesn't care about the phones. If you phones are behind nat use nat=yes > > for each defined account. > > The fix is incorrect. Asterisk chan_sip.c must distinguish between > SIP peers that are behind the firewall (together with the *) and those > that are on the outside. Either the configuration flag use_extern_ip > must be specific to a peer, or it must be figured out in some other > way. A global variable won't do since it creates a situation where > either external or internal peers will work but not both. Internals can use the IP address of the NAT box as the Asterisk Server IP and then it should work. i.e. don't set your internal SIP UAs to connect to the internal IP address of the Asterisk Server. The fix allows asterisk to work together with the NAT box to appear to all concerned as if it has a real IP address. cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recommended places for beginner to start?
On Sun, Nov 02, 2003 at 05:56:02PM -0700, Matthew England wrote: > (The list may get this msg twice; I originally sent it from the wrong email > address, my apologies. Moderator, if you can, please delete my original email > submission from [EMAIL PROTECTED] Thanks.) > > Hello- > > Summary: > > Can anyone recommend a place to start to learn how to create an Asterisk system > given a basic Digium PCI card and some generic PC hardware? I started with: http://www.automated.it/guidetoasterisk.htm probably low enough level for someone coming from a Windows admin background, perfect for someone who has been in "Marketing Land" (anyone read Dilbert? :-) for a while. Enough to get up a demo system. cheers, Woody > Details: > > I plan to help a friend not familiar with Linux platforms load and evaluate a > Digium/Asterisk system for business-development purposes. A couple years ago I > used to work as a Unix/Linux sw developer and sysadmin, but have been doing > sales/marketing stuff since. > > Where should I start to read about loading a system? My friend apparently has > a $100-flavor of Digium for eval purposes (can hook up to one external phone > line, or so I'm told), but knows little else. Since I've been the unix/linux > geek in a past life, he came to me for assistance. > > I downloaded the .pdf handbook, and their appeared to be a reference to a > "downloading and installing" section, but I couldn't find any text/body that > actual described this process. > > Do I pick any linux flavor (presumably with compatible kernel) like RedHat/ > Debian/SuSE and load up the source/pkgs/rpms necessary and let 'er rip? Will I > get a phone switch/PBX (or whatever this is) going fairly easily, assuming I > get my linux box/platform fired up ok? Any gotchyas, tricks of the trade, > things to know/worry about, etc? Is this all contained in the .pdf handbook? > When I skimmed it, I didn't find anything that seemed to match up with a > "installing for a rookie's" perspective like mine, but maybe I overlooked > something. > > I have yet to get my hands on the Digium hardware/docs/etc that my buddy > ordered; maybe some answers/secrets/support-resources are in there? > > I'm on vacation right now and am a little short on info, but before delving > into this when I get back (probably starting around 11/5) I thought I would > send out this note to the user list so that I might potentially save some time > in research/pain before I start. > > Thanks for any help! > -Matt ___ Asterisk-Users mailing > list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/ > asterisk-users -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] msn messenger
On Sat, Nov 01, 2003 at 09:35:26AM +0100, Florian Overkamp wrote: > At 01:43 1-11-2003 +0300, you wrote: > > >Is msn messenger capable of using asterisk as it's gateway? > > Yes, provided you are using MSN 4.7, and not 5.0 or higher. Configure the > Communications Service under the Options/Accounts pane. I'm not sure either way with MSN Messenger, but Windows Messenger (slightly different - same servers - no adds) which comes with XP does, there is a registry key (HKEY_CURRENT_USER->software->messenger_service->corpPC2phone) which you need to change to '1' to get the "make a phone call" link down the bottom of the window. Maybe there is a Wiki article about setting up softphones? cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
On Thu, Oct 30, 2003 at 07:29:17PM +, Gavin Hamill wrote: > On Thu, Oct 30, 2003 at 03:10:09PM -, Linus Surguy wrote: > > > One option you might not have considered is connect your existing PBX > > to the back of Asterisk and thereby use it as a channel bank itself. > > Very interesting :) > > There *is* an 'S-bus' (which is the same as an 'S0-bus'?) I'm told, > which we run 4 faxmodems off - I'm not exactly sure /how/ they connect, > tbh.. will need to check that out... Perhaps they're just 4 POTS > analogue extensions... S-bus might be ISDN BRI ports, in which case Asterisk can plug in with an AVM Fritz (~110 euro) and chan_capi. > This would be the ideal testing ground for Asterisk (for me to learn on) > since hopefully we could pass the incoming number to the S0-bus, hence > Asterisk, hence any IP Phones we buy as a technology demo. > > The idea of taking a fresh ISDN30 and trying to get everything working > from day 1 terrifies me :) > > We've looked at 'myPBX' from > http://www.telappliant.net/site2/mypbx_solution.htm > > And whilst I like the idea of a pre-configured appliance, I don't know > if you get root access, etc. since we will need to write our own > applications, etc. AGI (asterisk gateway interface??) is an application interface for Asterisk, which can use perl, C, php and probably other languages... > As always, I'm open to ideas =) A good philosophy. cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
On Thu, Oct 30, 2003 at 03:10:09PM -, Linus Surguy wrote: > > > > Finally, are my options for handsets limited to IP phones via Ethernet, > > or analogue phones via a channel bank (and then to another Digium E1/T1 > > card), or is there the possibilty to re-use proprietary handsets from a > > previous PBX? > > One option you might not have considered is connect your existing PBX to the > back of Asterisk and thereby use it as a channel bank itself. And the reverse is possible too, if you buy 2 E100 cards, you can plug your old PABX into the Asterisk server and set up with very minimal config so each proprietry handset can be used with Asterisk. Of course you only get 30 simultaneous connections between the Asterisk and the old PABX per E1 If you are going this route, you should consider a TE410P which will give you future options of T1 channel banks, extra E1 lines etc. -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicetronix OpenLine4
On Mon, Oct 27, 2003 at 04:34:38PM -0500, Jorge Mendoza wrote: > Hi, > > I have two OpenLine4 boards, and would like to test with *. > But I see in vpb.conf that only V6PCI/V12PCI is mentioned. It means that > OpenLine4 does not work? The author of chan_vpb.c was expecting a patch from someone who had it working when I contacted him about 6 weeks ago, so it has probably progressed further than that since then. cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk config files
On Fri, Oct 24, 2003 at 08:32:46AM -0500, Todd Wallace wrote: > Would anyone mind sending me a working set of config files for asterisk and > their softphone settings? I am really looking for very basic setup stuff as I This is real basic I have the make sample from Oct 14. I added the following to sip.conf: [woody] type=friend insecure=yes username=woody secret=bogus host=dynamic defaultip=192.168.2.76 [pyro] type=friend insecure=yes username=pyro secret=bogus host=dynamic defaultip=192.168.2.243 and the following to extensions.conf: exten => 1976,1,Dial(SIP/woody,15,tr) exten => 1974,1,Dial(SIP/pyro,15,tr) and then set up windows messenger (tools..options..accounts..Communications Server Account..Advanced) to connect to the asterisk using UDP only with the sign-on names "[EMAIL PROTECTED]" and "[EMAIL PROTECTED]", usernames "woody" and "pyro", and passwords "bogus". Look through extensions.conf for numbers you can call from messenger. Oh, also you'd need to change your windows registry HKEY_CURRENT_USER..software..messenger service..corpPCphone from 0 to 1. The "Make a phone call" comes up under your "I want to..." menu at the bottom of your messenger window. > just want to show that the system works to my management before they will allow > me to spend the money on phones and a telco card. Server is a redhat 7.3 w/ > 512 RAM and dual 550's. Asterisk tar ball has been laid down, configured, > make, make install, and make samples done... > > email address: [EMAIL PROTECTED] > > > Todd Wallace -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New here...
On Fri, Oct 24, 2003 at 12:28:39AM -0500, TODD WALLACE - Mail Lists wrote: > I am trying to get an initial setup up and going which I assume is a very > common question here. My basic questions are the following: > > Can I get Asterisk up and going without voice cards using it with SoftPhones > internally as a proof of concept. (just calling extensions and leaving voice > mail) Yes. > Is there a jump start config that would accomplish this? The last step in installing asterisk is "make install". After that, if you type "make samples" you get the samples configs put in, which lets you leave voicemails, call digium via IAX, etc. http://www.automated.it/guidetoasterisk.htm Is a very nice intro. > What is the recommended SoftPhone that is "Open Source" ? Open Source I guess it would be GnoPhone Free: Windows Messenger (not MSN Messenger) X-Lite NetMeeting cheers, -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Australian Options
On Thu, Oct 23, 2003 at 01:05:41PM +1000, Matthew Enger wrote: > Hello, > > Is A-tick just a certification or is something done to each card? I contacted It is two parts. 1) A certification 2) A sticker which goes on the card > the Australian distributor and their wholesale prices are twice the price of > that of a UK supplier who frequents this list! So I am wondering what does > A-tick mean? A-tick means that you can use the device with Australian Telcos without written permission. Connecting a non A-tick device to the Telco carries a fine of about $20,000. Getting something A-tick certified is expensive ($10,000 +) so I think the distributor should be repaid for that and rewarded for the risk he took in getting it approved before he had the sales to pay for it. Anything which can be plugged into a telecomms network needs to be A-ticked or "A-crossed" (marked in a special way that it can't be plugged into the telecomms network) before it can be sold here. If you get the card from the UK, I don't think it's covered by the A-tick work done by the Distributor, so you are risking a huge fine. That's my understanding anywaay. cheers, Woody > > Thanks, > > Matthew Enger > [EMAIL PROTECTED] > > Quoting Anthony Wood <[EMAIL PROTECTED]>: > > > On Tue, Jul 29, 2003 at 12:28:48AM +1000, Adam Goryachev wrote: > > > > Adam why would you need a channel Bank for the E100P? > > > > > > > > Regards > > > > > > > > Mark McKibbin > > > > > > As I specified below: > > > > > > > Again, with smaller offices (say 15 or less extensions) the price of a > > > > TE400P card is too expensive, will there be a TE100P card produced and > > > > approved for use in Australia? > > > > Even if it is, will this solve the problem? I'm thinking not, because > > > > the single port would need to connect to a channel bank which in turn > > > > would connect to the telco plus extensions, therefore we are back to > > > > the same situation as with analog lines above. > > > [ie, requiring an E1 channel bank with approval for use in Australia] > > > > > > So, I still haven't received a single suggestion to this question: > > > > > > > So, really, are there any decent solutions for asterisk use in > > > > Australia? > > > > > > > > PS, currently I am using the TDM400B plus Traverse Netjet ISDN with I4L, > > > > but this doesn't seem suitable quality for re-sale. > > > > > > Add-on, I know this isn't suitable quality for re-sale because it uses I4L > > > and not chan_capi, but I don't know of any chan_capi ISDN hardware which > > is > > > approved for use in Australia either. > > > > (yes this thread is 3 months old) > > > > The Australian distributor for AVM has AVM Fritz PCI cards for sale which are > > A-ticked. > > > > The price is steeper due to the cost of the A-tick. > > > > cheers, > > Woody > > > > > If there was such a thing, this would at least easily enough provide 4 > > > channels + 8 extensions with 2 x ISDN + 2 x TDM400B cards Then of > > > course, for 10 channels in with 14 extensions ?? Still need a channel bank > > > solution, or force the use of an OnRamp 10 with the TE400P which doesn't > > > have approval yet. > > > > > > Regards, > > > Adam > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > > Woody > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > This message was sent using IMP, the Internet Messaging Program. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Australian Options
On Tue, Jul 29, 2003 at 12:28:48AM +1000, Adam Goryachev wrote: > > Adam why would you need a channel Bank for the E100P? > > > > Regards > > > > Mark McKibbin > > As I specified below: > > > Again, with smaller offices (say 15 or less extensions) the price of a > > TE400P card is too expensive, will there be a TE100P card produced and > > approved for use in Australia? > > Even if it is, will this solve the problem? I'm thinking not, because > > the single port would need to connect to a channel bank which in turn > > would connect to the telco plus extensions, therefore we are back to > > the same situation as with analog lines above. > [ie, requiring an E1 channel bank with approval for use in Australia] > > So, I still haven't received a single suggestion to this question: > > > So, really, are there any decent solutions for asterisk use in > > Australia? > > > > PS, currently I am using the TDM400B plus Traverse Netjet ISDN with I4L, > > but this doesn't seem suitable quality for re-sale. > > Add-on, I know this isn't suitable quality for re-sale because it uses I4L > and not chan_capi, but I don't know of any chan_capi ISDN hardware which is > approved for use in Australia either. (yes this thread is 3 months old) The Australian distributor for AVM has AVM Fritz PCI cards for sale which are A-ticked. The price is steeper due to the cost of the A-tick. cheers, Woody > If there was such a thing, this would at least easily enough provide 4 > channels + 8 extensions with 2 x ISDN + 2 x TDM400B cards Then of > course, for 10 channels in with 14 extensions ?? Still need a channel bank > solution, or force the use of an OnRamp 10 with the TE400P which doesn't > have approval yet. > > Regards, > Adam > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] e100p in Australia
On Thu, Oct 16, 2003 at 12:28:48AM +1000, Michael A. Miller wrote: > Just a couple of notes on the ISDN issue in Australia... > > I agree with the audio quality of the NETjet-s. I also had a horrible time > working out the DTMF detection under ISDN4Linux. Overall, I can not fault > the hardware and guys at Traverse were quick to help out but it is not > production quality for audio. > > I finally had to switch to the AVM FRITZ!Card PCI. It is A-Tick approved. > The local distributor is listed on their website. The audio quality is > great and there are no DTMF issues with the card. > > Please feel free to let me know if anyone would more info on my > experiences with both cards. I'm keen to hear the Fritz! in action, or details on feedback/experiences/features which work/etc. cheers, Woody > > > Michael > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Wood > Sent: Wednesday, October 15, 2003 5:01 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] e100p in Australia > > On Wed, Oct 15, 2003 at 12:50:33PM -0500, Stephen Dredge wrote: > > > > I've seen this question asked before but haven't seen a definative > answer. > > Does the e100p work in australia? Did any one who was asking the > question > > You need written permission from your Telco to use non-approved hardware, > I think US $595 + AU $10,000 fine is more than US$1495 :-) > > > before bite the bullet and get one? I can get a te410 if i really have > to but > > would prefer to stay with the cheaper option. > > NetJet-S apparently works with ISDN BRI (E.g. onramp 2) and Asterisk, but > there > are some echo issues for the local end of the call. > > There was a Melbourne guy using one, I'm not sure how he's gone in the > last 4 months: > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Mark > > McKibbin > > Sent: Friday, 13 June 2003 1:01 p.m. > > To: [EMAIL PROTECTED] > > Subject: [Asterisk-Users] E1 cards > > > > > > We are not having any luck with the E100p card here in Australia, it > > will work with a crossover cable to another device but will not talk to > > our Telco Telstra who probably have a weird implementation of an E1. > > > > Any suggestions on a replacement? > > > > Regards > > > > Mark McKibbin > > DCS Internet > > 64 Queen St > > Warragul > > Victoria3820 > > Australia > > www.dcsi.net.au > > [EMAIL PROTECTED] > > Ph. 1300 665575 > > Fx. 1300 556595 > -- > Woody > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] e100p in Australia
On Wed, Oct 15, 2003 at 12:50:33PM -0500, Stephen Dredge wrote: > > I've seen this question asked before but haven't seen a definative answer. > Does the e100p work in australia? Did any one who was asking the question You need written permission from your Telco to use non-approved hardware, I think US $595 + AU $10,000 fine is more than US$1495 :-) > before bite the bullet and get one? I can get a te410 if i really have to but > would prefer to stay with the cheaper option. NetJet-S apparently works with ISDN BRI (E.g. onramp 2) and Asterisk, but there are some echo issues for the local end of the call. There was a Melbourne guy using one, I'm not sure how he's gone in the last 4 months: > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Mark > McKibbin > Sent: Friday, 13 June 2003 1:01 p.m. > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] E1 cards > > > We are not having any luck with the E100p card here in Australia, it > will work with a crossover cable to another device but will not talk to > our Telco Telstra who probably have a weird implementation of an E1. > > Any suggestions on a replacement? > > Regards > > Mark McKibbin > DCS Internet > 64 Queen St > Warragul > Victoria3820 > Australia > www.dcsi.net.au > [EMAIL PROTECTED] > Ph. 1300 665575 > Fx. 1300 556595 -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Eicon Diva Server BRI (T1) Cards
On Wed, Oct 15, 2003 at 12:02:18AM +0200, Roger Schreiter wrote: > Hi, > > my asterisk experiences with isdn cards supported by i4l Out of interest, which cards. > are not very good, but with avm a1 and capi everything > works very fine and stable. (SuSE Linx 8.2, Kernel 2.4.20, > german ISDN). Heard good things about capi and AVM, but unfortunately none are approved for Australia :-( > Now I want to connect a T1. Should I use an AVM T1-B > for approx 6000 EUR or is it ok to use one of Eicon's cheaper > Diva Server BRI S2M cards? T1 = ISDN PRI in the US/Japan (24 lines) ISDN BRI = 2 lines So I don't think that card would work, you want a US PRI card if they make one. If you are in Germany, I think you want E1 = ISDN PRI outside US/Japan (30 lines) and for that you want a E100P for US$595 from digium. > Do they use i4l or is there a similar thing as AVM's capi > driver for eicon's S2M cards? I think there is a channel driver for the eicon, but I don't know where from. > Are there any experiences how they compare when used > with asterisk? I am interested in this too, the Eicon cards are also certified for Australia. cheers, -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] out going calls
On Tue, Oct 14, 2003 at 08:37:29AM +0930, [EMAIL PROTECTED] wrote: > I am not having any luck placing out going calls > > I dial the number 08 82420173 ( our outside line ) Hi Mick, calling your outside line would be engaged anyway, if you are using it to call out on, no? Call your mobile instead. I'm interested with how you go with Voicetronix, they are one of the few Australian approved cards. cheers, Woody > But all I get is engaged signal and log this. > > Oct 14 08:40:14 DEBUG[16401]: File pbx_wilcalu.c, Line 65 (autodial): > Entered Wil-Calu fd=20 > Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 657 (create_addr): > Setting NAT on RTP to 0 > Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 548 (__sip_ack): > Stopping retransmission on > '[EMAIL PROTECTED]' of Request 102: Found > Oct 14 08:40:31 DEBUG[8201]: File chan_sip.c, Line 3841 (check_user): > Setting NAT on RTP to 0 > Oct 14 08:40:31 DEBUG[8201]: File chan_sip.c, Line 548 (__sip_ack): > Stopping retransmission on > '[EMAIL PROTECTED]' of Response 101: > Found > Oct 14 08:40:31 DEBUG[8201]: File chan_sip.c, Line 3841 (check_user): > Setting NAT on RTP to 0 > Oct 14 08:40:31 DEBUG[8201]: File chan_sip.c, Line 4891 > (handle_request): Check for res for mick > Oct 14 08:40:31 DEBUG[8201]: File chan_sip.c, Line 973 (find_user): Call > from user 'mick' is 1 out of 0 > Oct 14 08:40:31 DEBUG[8201]: File chan_sip.c, Line 548 (__sip_ack): > Stopping retransmission on > '[EMAIL PROTECTED]' of Response 102: Not > Found > > > > Any ideas would be appreciated > > Regards Mick West > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] echo for 15 seconds
On Thu, Sep 25, 2003 at 11:56:04AM +1000, Shaun Ewing wrote: > > - Original Message - > From: Chad R. Graham > > >For the first 15 seconds of a call I get echo on the ata 186 side only. I > >assume after that the echo canceller kicks in but is there any way to make > it > >happen faster? > > Same thing here - except we're using Cisco 7960 and 7940 IP phones. > > We're getting used to it, the main thing is that the remote caller doesn't > hear it (which they don't). A person visiting our office and using the phone > may get a bit of a surprise though. > > Incoming calls it's not an issue, as the caller would have already gone > through the IVR/auto attendant and presumably the echo cancellation has > already kicked in before reaching the operator. > > A related question for those reading this post - is echo cancellation an > issue with ISDN? I've thought of converting the POTS lines to a couple of > BRI lines - but if echo is still a problem then I'll leave it until we > actually need the extra lines. I've heard from a user on this list that: 1. there is some echo using ISDN4Linux 2. chan_capi removes the echo 3. you can only use one card with chan_capi Others on the list can correct me/them, I want to know the real story... cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP / GrandStream Configuration
On Thu, Sep 25, 2003 at 12:33:02AM -0400, Uriel Carrasquilla wrote: > Adam: > I believe you. I assume that the RTP is creating a symetric configuration > between * and the SIP phone. The situation we are left to live with is that > * (won't be the Sip phone) can only live in the Internet brave world (and > not behind a firewall). is this acceptable? > Uriel You could set up a tunnel between both NATed networks so they didn't need to use NAT. Assuming there are no IP conflicts. This will add some latency, I'm not sure how much. cheers, Woody > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Stephen Varga > Sent: Wednesday, September 24, 2003 11:02 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] SIP / GrandStream Configuration > > > On Wed, 2003-09-24 at 21:50, Uriel Carrasquilla wrote: > > Adam: > > in reference to my first message, the NAT on the SIP/GS (a D-Link router) > > has ports 5060 for SIP-registration and RTP ports 5000 to 5008 being > > forwarded to the Sip/GS. > > The Asterisk server, also behind another NAT (Linksys), has the same ports > > opened and forwarded. > > is it still impossible? > > URiel > > Nope, it is not currently possible. * behind a NAT for SIP does not work > because the * real IP address is placed in the SDP information, > therefore the 'outside' phone can not send the media stream to *. See my > answers over the last week for the more details and possible work > arounds. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI CallerID problem
On Wed, Aug 20, 2003 at 12:39:49PM -0700, Michael Rose wrote:
> Greetings all..
>
> We have an inbound/outbound PRI installed and terminated on a T400P –
> Digium Quad T1 card. We’re seeing an odd problem when sending
> $CALLERIDNUM when calls from the PRI are forwarded back out to the PSTN
> over the PRI. The $CALLERIDNUM is not being sent out along with the
> call. It’s sending the phone number of the PRI itself, rather than the
> $CALLERIDNUM information.
Is this the scenario you are describing?
You call the office from home, the direct number of a collegue.
He is not in, asterisk forwards it to his mobile.
His mobile rings and gives the PRI number as the caller ID, but you want it to give
out your home number?
>
> Yes, we can send CID info to our PRI provider. If we make a call with
> our Cisco 7960, we can send any phone number we enter into
Even numbers not associated with your PRI?
If you can, then your provider is very trusting, letting you spoof caller ID.
Perhaps your provider will only let you set caller ID to a number in your range of
numbers.
Otherwise, perhaps the provider of the called parties line will only let you provider
send caller id from it's numbers, and your home number (in my example) is not
part of that.
cheers,
Woody
> SetCallerID(). However, if we use SetCallerID(${CALLERIDNUM}) it won’t
> forward the CID number.
>
> [mydid]
> exten => 5558384810,1,SetCallerID(${CALLERIDNUM})
> exten =>
> 5558384810,2,Dial(Zap/g1/15553456131&SIP/[EMAIL PROTECTED]&SIP/[EMAIL PROTECTED],20)
> exten => 2068384810,3,Congestion
>
> Now when I call my DID, 5558384810 from my land line, Asterisk takes the
> CID number and sends it to my IP phones (Cisco 7960) which I can see on
> the display of the IP Phone. However, the CID number doesn’t get sent to
> the cell phone.
>
>
> Below is the context we use to make outgoing calls from our Cisco
> 7960’s. This works fine with our PRI. When I call using the following,
> my cell phone shows the incoming call coming from 2065551212. This tells
> me that our PRI vendor is allowing us to send CID info.
>
> [dialout-pri]
> exten => _1NX,1,SetCallerID(2065551212)
> exten => _1NX,2,Dial(Zap/g1/${EXTEN},100,T)
> exten => _1NX,3,Congestion
> exten => _1NX,4,Hangup
>
>
> Here’s what Asterisk shows, when the call is coming in.
>
>-- Executing SetCallerID("Zap/23-1", "5557209085") in new stack
> -- Executing Dial("Zap/23-1",
> "Zap/g1/15557206131&SIP/[EMAIL PROTECTED]&SIP/[EMAIL PROTECTED]|20") in new
> stack
> -- Called g1/15557206131
> -- Called [EMAIL PROTECTED]
> -- Called [EMAIL PROTECTED]
> -- Accepting call from '5557209085' to '5558384810' on channel 23,
> span 1
> -- SIP/69.28.200.84-4698 is ringing
> -- SIP/10.0.1.2-6695 is ringing
> -- Zap/1-1 is ringing
>
>
> Thanks in advance.
>
>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
--
Woody
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware question
Here are some options: Digium X100P x 4 US$100 * 4 = US$400 well supported by asterisk manufacturer supports asterisk developers Deployed in lots of places with Asterisk Voicetronix OpenLine4 - US$500? Use 1 PCI card reported working with chan_vpb manufacturer supports linux Can be used with other software Voicetronix OpenSwitch 6(12) US$750?(US$1800?) Use 1 PCI card works with chan_vpb manufacturer supports linux Can be used with other software 6/12 ports can be jumpered in pairs(4s?) to be FXO or FXS. Channel Bank + T/E 100/400 P US$? use 1 PCI card On Wed, Aug 20, 2003 at 08:27:06AM -0700, Bruce Ferrell wrote: > Digium makes a 4 port card. It'd be hard to get 4 lines with quicknet > hardware. > > Bartosz Jozwiak wrote: > > Hello, > > > > Again one more question about hardware. > > What could you suggest me to buy. > > I need hardware to connect let's say 4 analog lines. (FXO). > > This hardware should "talk" to Asterisk of course.. > > Thanks very much for some advices :) > > > > Bartek > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Get faxed you faxing faxer!
On Tue, Aug 12, 2003 at 11:48:31AM +1000, Adam Donnison wrote: > Fax is not as easy as it may first appear. First of all there > are two parts to the protocol, one of which would be relatively > easy to accomplish if you have built your softmodem as it uses > flow-controlled error detecting style conversations. But this > is just the call setup, and inter-page negotiation. The > actual sending of the page data uses a non-flow controlled > protocol that uses bit stuffing to make sure the receiving fax > receives a line of data just in time to print it. The timings > are _very_ tight, from memory all you need is a delay of well > under 100ms for the remote machine to give up. Even with > TR29.2 modems (where the T.4 data transfer protocol timings > are handled by the modem) you can get all sorts of problems unless > you can make sure you keep the modem buffer full. Doing it > all in software means you have to manage the timings yourself. > > From my perspective, unless you are using a real-time OS, you > may as well forget implementing T.4 as your interrupt latency > is likely to kill you. Either that or using a SMP machine > so that there is always a CPU available. Either way, it is > a big ask. I concur. Even hardware Class 1 faxmodems require a lot of timing dependant stuff to happen on the server, class 2 and 2.0 can handle most of it themselves. My experience is with hylafax, you really want dedicated hardware (faxcard,faxmodem class 2, or dedicated box), to handle the fax unless you want a successful fax reception to be the exception, rather than the rule. Maybe if you only got faxes at night and only got asterisk calls during the day, it would be OK, but it is amazingly annoying that a $20 second hand 15 year old thermal fax machine recieves every fax perfectly and hylafax still has hiccups now and then. cheers, Woody > > Adam > > Eric Wieling wrote: > > software fax modem dsp > > > > There doesn't seem to be many software fax modems out there, but I found > > these few when doing a Google search a few months ago and bookmarked > > them: > > > > http://www.softmodem.org/ > > > > http://fabrice.bellard.free.fr/linmodem.html > > > > http://artax.karlin.mff.cuni.cz/~pacoo0bm/archive/new_fischer/ > > > > http://www-users.cs.york.ac.uk/~fisher/modem/ > > > > I just realized that softmodem.org links to all the other URLs. > > > > On Mon, 2003-08-11 at 19:23, Jamie Carl wrote: > > > >>All this talk of faxing has started me thinking (this is > >>always a bad thing) and I've come up with a question. > >> > >>Now, I know Asterisk can detect and route faxes, to a > >>'fax' extension and all that. But can Asterisk be used to > >>'receive' faxes? > >> > >>I know there was some talk about this just over a year ago > >>and I'm wondering if anything came of it. It would be > >>nice to be able to receive faxes with asterisk and then > >>have them filer through and AGI script and emailed or > >>dumped in a directory/database/whatever. > >> > >>I mean for me, currently, I have a fax/modem hooked up to > >>the same machine asterisk is on, and it get's sent faxes > >>via my FXS port. It would be nice to be able to pull off > >>the modem and throw it away. (or auction it on ebay) > >> > >> > >> -- > >>BTEL Consulting > >>850-484-4535 x2111 (Office) > >>504-595-3916 x2111 (Experimental) > >>877-552-0838 (Backup Phone) > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > Adam Donnison email: [EMAIL PROTECTED] > Saki Computer Services Pty. Ltd. > 93 Kallista-Emerald Roadphone: +61 3 9752 1512 > THE PATCH VIC 3792AUSTRALIAfax: +61 3 9752 1098 > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why are FXO so expensive?
On Tue, Aug 05, 2003 at 11:07:22AM -0500, Steven Critchfield wrote: > On Tue, 2003-08-05 at 06:59, Samy Touati wrote: > > Hi, > > > > I've been browsing for FXO devices, and I'm really surprised at their > > costs. > > Why such devices are so expensive and somehow hard to get ? > > Because most people that would need the items can quickly scale beyond > analog incoming lines. So there is not a glut of them on Ebay to drive > the prices down. Also FXO devices require certification to be used in many countries, and this is not cheap, the manufacturer has to pass this cost on to someone. But if you want 24 or 30 lines, a Digium Wildcard T100P/E100P is only $25/$20 per line or for 96/120 lines, a T400P/E400P is only $16/$13 per line. For smaller systems, you'd have to go a NetJet ISDN BRI card ($150? for two lines) or X100P ($100?) per 1 line, or VoiceTronix OpenLine4 ($500?), OpenSwitch 6/12 ($800/$1500?) for 4/6/12 analogue lines. The openswitch cards can operate groups of the ports as FXS ports too. cheers, Woody PS: $US ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Australian Options
On Tue, Jul 29, 2003 at 12:34:48PM +1000, Adam Goryachev wrote: > > Why don't you just plug the e100p into the onramp10-30 why do you need a > > channel bank? > > > > Regards > > > > Mark McKibbin > > How do you connect your extensions then? If you use the single T/E 1 port > for your onramp, you end up with nil extensions, using a channel bank you > can use the single T/E1 port for both 10 channels from the telco plus 14/20 > channels for extensions... > > BUT, you need an australian approved channel bank. I don't think you do. You need an Australian approved TE410, but anything past that doesn't interact directly with the telco's network, so doesn't need approval. So a 'yank bank' would be legal. Also, a TDM400 would be legal, because you aren't plugging that into the telco's network. > Or am I just being stupid, and there is something obvious I am missing for > the use of asterisk in Australia *legally* ?? You need something approved to plug into the telco network. At the moment, I think that means OpenLine4 or NetJet ISDN BRI. Things which don't plug into the telco don't need approval (e.g. FXS cards, IP Phones). > PS, for those interested, there is an option of the australian made > openswitch cards, but since they are unsupported for use with asterisk, I > don't want to go out and purchase them/sell them to a client only to find > that it doesn't quite work properly in some situations... Voicetronix has been known to let you trial them for a month or two (with a credit card slip) to see how they work out. I think they should be fine (no echo) if you are using hardware for everything, like it sounds you want to do. But the openswitch 12 isn't approved yet either AFAIK. cheers, Woody > PPS, anyone who is successfully using the openswitch cards with asterisk, I > would appreciate feedback on them Not yet. The developer of the asterisk driver says the OpenSwitch 12 works and has heard reports of the openline 4 working. > > Regards, > Adam > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AUSTEL Certified
On Wed, Jul 16, 2003 at 03:38:39PM +1000, Gary wrote: > PLEASE NOTE THIS. > > > NONE, I do mean none of the digium cards are legal to use in australia > as they have no c-tick or a-tick (each as required). IANAL, but it is pretty clear from the regs that you can plug your phone line into whatever, if you get written permission from the telco. I'm yet to ask. There are also exceptions for mobile phones imported for personal use and testing equipment. It's also unclear as to where the network boundary is for installations of Optus VoDSL, I'm trying to clear this up with ACA at the moment. AUSTEL was the old scheme, ACA runs the show now. I think the cheapest legal way to plug asterisk in at the moment is to upgrade your phone line to ISDN BRI ($125 upgrade from Telstra at the moment) and buy a NetJet ISDN card (AU$250). Does anyone know if these cards work with capi? cheers, Woody > > Under certain circumstance you may be able to use the E100p/E400p; to > hook up to a telco, but only under particular circumstances which would > also involve colocation. > > Now the TE410 does even exist yet, please read the thread... read > winter for summer for the USA being upsidedown if your from downunder > ;-) > > Gary > > On Wed, 16 Jul 2003 14:43:18 +1000, David Hindmarsh wrote: > > >Hi Guys > > > >Is this card available here in Australia yet, is it certified. > > > >Dave > > > >-Original Message- > >From: [EMAIL PROTECTED] > >[mailto:[EMAIL PROTECTED] Behalf Of Steve > >Underwood > >Sent: Monday, 14 July 2003 1:29 AM > >To: [EMAIL PROTECTED] > >Subject: Re: [Asterisk-Users] AUSTEL Certified > > > > > >Rainer Jochem wrote: > > > >>>We plan to have certification on the new TE410P board by the end of > >>>summer. > >>> > >>> > >>TE410P? > >>*getting curious* > >> > >>What kind of board will this be? > >> > >4 x T1/E1 ports, bus mastering. One board does T1 or E1 under software > >control. I don't know if you can mix E1 and T1 modes on different ports, > >though. > > > >>We are currently looking for PRI interfaces for our asterisk-box - so > >>perhaps we'll wait a bit more (depending on what "end of summer" means ; > >> > >About April in Australia :-) > > > >Regards, > >Steve > > > > > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > >--- > >Incoming mail is certified Virus Free. > >Checked by AVG anti-virus system (http://www.grisoft.com). > >Version: 6.0.497 / Virus Database: 296 - Release Date: 4/07/2003 > > > >--- > >Outgoing mail is certified Virus Free. > >Checked by AVG anti-virus system (http://www.grisoft.com). > >Version: 6.0.501 / Virus Database: 299 - Release Date: 14/07/2003 > > > > > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > > . > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: X100P mod or USB relay box, RE: [Asterisk-Users] Line Override Device
The Voicetronix Openline6 and Openline12 cards have the functionality you want built in. You can configure (jumpers) which ports are FXO and which are FXS (in groups of 2 IIRC) and 1st FXO goes to 1st FXS etc. in case of power failure. Apparently these cards work with Asterisk (chan_vpb). I think cost is AU$1500 and AU$3000 for 6 and 12. cheers, Wooody On Mon, Jul 14, 2003 at 02:17:59PM -0400, Reed Wade wrote: > > > At 12:57 PM 7/14/2003 -0500, you wrote: > >This makes me think that you could take this a step further too and > >incorporate an external power supply and a relay that could interupt > >mains power so that you could power cycle the PC if the watchdog had > >power to operate and the PC wasn't responding or generating pings. > > > i like that > > -reed > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digital phones
On Sun, Jul 06, 2003 at 09:42:55PM -0400, marrandy wrote: > On Sunday 06 July 2003 04:26 pm, Steven Critchfield wrote: > > > The problem isn't as much controlling the phone, but electrically > > interfacing with it. AFAIK, this is one of the ways the proprietary > > systems lock you into their phones. Currently there is no hardware to > > talk to those phones, let alone the programming to do it. If you find a > > card that supports those phones and plugs into a PCI bus, hopefully also > > already has linux drivers, then we can see about supporting those > > phones. > > -- > > Steven Critchfield <[EMAIL PROTECTED]> > > > So does that just leave regular single line phones ? > > Besides IP phones. > > What else can be used ? You can plug your old PABX into asterisk and use its phones through that. Softphones (software + computer + soundcard + microphone + speaker) Some IP phones have bluetooth gateways so you can use your snazzy mobile phone without paying your telco. cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk vs. system user accounts
No-one has mentioned PAM yet. (pluggable authentication modules). If you implement PAM in Asterisk, then you have LDAP/passwd shadow windows etc. in one step. Maybe the phone numbers in /etc/passwd will get used! cheers, Woody On Tue, Jun 24, 2003 at 04:17:15PM -0500, Steven Critchfield wrote: > On Tue, 2003-06-24 at 15:47, Dylan VanHerpen wrote: > > After saying LDAP is a better choice than system users, I still wonder > > why it is important to have users be able to change passwords here. > > > > It would greatly simplify unified messaging: one account, all your > > messages (email, voice, fax) in one mailbox. > > This doesn't mean there needs to be a real account on the system. Exim > can deliver to non login accounts, Cyrus IMAP doesn't need real user > accounts, and I forget the pop3 daemon that works the same way. Hylafax > doesn't need a user to send you the faxes. So again, you only need some > simple way of keeping user/password mappings straight. LDAP would be > okay, if it was easy enough to setup and use. Not to mention for your > SIP users, LDAP can be used as a phone directory. I hope this helps you > move forward with something usable and secure. > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is this possible:
On Mon, Jun 23, 2003 at 04:07:32PM +1200, Aaron Martin wrote: > The hardware we are planning to use is: > > Micronet SP5050 FXO Gateway > http://www.micronet.com.tw/Products/VoIP/SP5050.asp > > Micronet SP5100 IP Phone > http://www.micronet.com.tw/Products/VoIP/SP5100.asp > I can't help you with specific hardware. Asterisk speaks SIP, MCGP, H.323, IAX. Both the products above speak H.323, so you should be able to hook asterisk in the middle. I guess you'd configure asterisk to register with the gateway, and the phones to register with asterisk. > We are hoping to use this hardware along with AsteriskPBX to replace our aging > PBX system. > > What I want to acheive is: > * Any incoming call from PSTN (via gateway) rings on the receptionists phone > for 10 seconds, and then starts ringing on all phones. > * Any user can just dial 1 to get the first avaliable outgoing line onto the > PSTN. > * All users can have voicemail, with email notification. > * Calls can be transferred from one user to another. > > Is this all possible?? Is anyone else using this Micronet hardware? Asterisk can do all the above with other hardware. Micronet should work in theory. Perhaps someone else on the list has had experience with the micronet hardware. cheers, -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P questions
On Mon, Jun 16, 2003 at 05:10:33PM -1000, Matthew John Darnell wrote: > > > > > Is there a product to bring in more than one POTS line short of a full > > > T1? It just seems silly that the technology hasn't advanced any > > > further than to have a single line per card. > > > > We are working on an FXO module for the TDM400P and hope to have it ready > > in a couple of months for initial testing, but no firm deadline. When > > that's ready we will have a multiport solution. In the mean time, there > > is always VoiceTronix, which now has support within Asterisk contributed. > > Does that mean that the Voicetronix is navibly supported by Asterisk? dict navibly | cat No definitions found for "navibly" Natively? All I know is: 1) There is a asterisk/channels/chan_vpb.c and asterisk/configs/vpb.conf.sample in the cvs. They seem to imply that openswitch12 and openswitch6 are supported. It took me a while to form the link vpb <=> voicetronix. 2) I got an email from the author of that file today saying that he had reports from another asterisk user that they had got it working with an openline4, and he was expecting a patch. 3) http://www.file-sure.com/ seems to be selling services related to asterisk and say voicetronix is supported. cheers, -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P questions
On Mon, Jun 16, 2003 at 12:39:20AM -0500, Asterisk wrote: > Hello! > > I've been following this list for several weeks now and would like to > purchase some hardware for a VOIP/ voice-mail solution. This card appears > strikingly similar to a modem. Is it? Is there a product to bring in more Well it has an RJ12 port and plugs into your computer and is green and has some chips and stuff on it, it is very similar looking :-) I think it is very similar in hardware and concept to a winmodem/linuxmodem/softwaremodem. > than one POTS line short of a full T1? It just seems silly that the > technology hasn't advanced any further than to have a single line per card. Well there are some products which may fit your bill, but will take more of your bills: www.voicetronix.com sells a 4 port PCI FXO at about AU$1000, a 6 port FXO/FXS (jumper selectable) at about AU$1500, and a 12 port PCI FXO/FXS (jumpers) at about AU$3000. (AU$1.50 =~ US$1) Also Dialogic: Dialogic D/41JCT-LS 4-port analog + voice - US$882.29 VFX/41JCT-LS 4-port analog + voice + fax - ? D/120JCT-LS 12-port analog + voice - US$1500 ebay cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E1 cards
On Fri, Jun 13, 2003 at 01:48:13PM +1200, Peter Armstrong wrote: > You need to get the ETSI or Euro version of PRA from Telstra and then it > will work, they offer it as well as their quaint version of PRA access. > > Peter Are the E100p cards Austel approved? Or does Telstra give you written permission? Or are you winging it :-) cheers, Woody > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Mark > McKibbin > Sent: Friday, 13 June 2003 1:01 p.m. > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] E1 cards > > > We are not having any luck with the E100p card here in Australia, it > will work with a crossover cable to another device but will not talk to > our Telco Telstra who probably have a weird implementation of an E1. > > Any suggestions on a replacement? > > Regards > > Mark McKibbin > DCS Internet > 64 Queen St > Warragul > Victoria3820 > Australia > www.dcsi.net.au > [EMAIL PROTECTED] > Ph. 1300 665575 > Fx. 1300 556595 > > > -Original Message- > From: Simon J Mudd [mailto:[EMAIL PROTECTED] > Sent: Friday, 13 June 2003 7:10 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Asterisk logging questions > > [EMAIL PROTECTED] (Michael Manousos) writes: > > > > While logger.conf indicates that you can log to a file I can't see > > > _where_ the logging is sent to (which directory). Looking at the > > > source it > seems > > > that the directory used is specified in ast_config_AST_LOG_DIR but I > can't > > > see where this variable is defined. Help? > > > > In /etc/asterisk/asterisk.conf, section [directories]: astlogdir => > > <.> > > Ok. Thanks. Having installed all the sample conf files I'm slightly > overwhelmed looking for things. > > Simon > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PDA's over SIP channels on Asterisk
On Tue, Jun 10, 2003 at 09:59:03PM -0700, Robert Hajime Lanning wrote: > > > On Tue, Jun 10, 2003 at 02:10:56PM -0500, Steven Critchfield wrote: > >> On Tue, 2003-06-10 at 13:49, flickds wrote: > >> > Is it possible for two PDA's to communicate like telephones via SIP > >> channels > >> > on a PC running Asterisk? If that is possible, does there exist any > >> > applications that can be installed on a Zaurus 5600, which is a PDA > >> with an > >> > Xscale processor running on a Linux OS, that can essentially turn it > >> into a > >> > softphone? Thanks in advance for any input, > > http://www.thekompany.com/embedded/tkcphone/ > > > My SL-5500 manual says, > > > > I/O device > > stereo headphone jack (monaural audio input) > > > > The manual has no description of the electrical or physical dimensions > > of the jack. > > > > It is an 1/8" jack. The plug on my cell phone seems to be closer to > > 3/32". No high precision, nor metric, measuring tools on hand. > > The left headphone is also wired to be a microphone. You can actually > yell into the left headphone when recording. (Though, it would work > better if something a bit more sensitive were connected, like a > microphone.) > > You can, actually, just get a 1/8 -> 3/32 stereo adapter from RadioShack. > Then connect a handsfree headphone to it. Or you can get a 3.5mm stereo male to 3.5mm left + right female adapter. This will let you plug a microphone into one and headphones into the other. cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Opportunistic VoIP
This is an idea from FreeSWAN, which was implemented in the recently released version 1.0. Basically the idea is that FreeSWAN sites automatically encrypt traffic between them when possible, without having to set up the link ahead of time. How this works is: The sites publish some info in DNS. FreeSWAN gets some traffic destined for that site. - looks up the info in DNS - if the info is there: sets up an encrypted connection - if the info is missing: sets up a normal connection This is a feature which can be turned off. How does this apply to asterisk? Asterisk has a call destined for a PSTN number Looks up the number in a central location If it's there, then connect to the reported IAX/SIP/whatever connection over the internet if it's up/ping is good/hops is good/whatever. Otherwise connects through the PSTN. Points: saves money possible quality issues for VoIP over many internet hops this isn't as good as the FreeSWAN way as there is no logical mapping between PSTN and DNS -- therefore need a central location potential for abuse what would be a good spot for the central location? comments? -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice Modem + Soundcard Driver
On Wed, Jun 04, 2003 at 09:15:25AM +0300, Dan wrote: > Hi Woody, > > As it is a software modem, it can do full duplex. > . If it can do full duplex, then no-one has pointed out to me why a driver can't be written for it. > > Also, I don't think Asterisk has drivers for this sort of thing, > > as most of the hardware is half-duplex. > > When asterisk is started, I see something like this in the log: > > [chan_modem_bestdata.so] => (BestData (Conexant V.90 Chipset) VoiceModem > Driver) I guess it's a specific chipset for certain hardware modems. > and > [chan_phone.so] => (Linux Telephony API Support) > == Parsing '/etc/asterisk/phone.conf': == Parsing > '/etc/asterisk/phone.conf' I guess we'll have to have a look into the Linux Telephony API, perhaps it is only half-duplex. > Can those two be used for a voice modem? Probably if you have a Conexant V.90 Chipset. chan_phone I don't know. > There is a modem_bestdata.conf file somewhere? Don't know. > > Thanks, > Dan > > > - Original Message - > From: "Anthony Wood" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Wednesday, June 04, 2003 2:09 AM > Subject: Re: [Asterisk-Users] Voice Modem + Soundcard Driver > > > > On Tue, Jun 03, 2003 at 11:09:28AM +0300, Dan wrote: > > > Hi, > > > > > > I have an USB voice modem which does not need to be connected to the > > > soundcard in order to build an answering machine. > > > It appear in the system as another sound card. > > > I have tested a free Answering machine application and it works great. > > > The modem is very cheap (paid around 25 USD for it) and has linux > drivers, > > > including the source files: > > > http://www.smlink.com/products/SmartUSB56.htm for description > > > and > > > http://www.smlink.com/download/Linux/slmdm-2.7.14.tar.gz > > > for the linux driver, including source files. > > > > > > Can this be used as a FXO interface for Asterisk? > > > > The key is can the modem do full-duplex voice (sound in both directions > > at the same time). An answering machine is half-duplex: > > first, the sound is sent from the machine to the caller, > > then, the sound is sent from the caller to the machine. > > > > Also, I don't think Asterisk has drivers for this sort of thing, > > as most of the hardware is half-duplex. > > > > But I am only a newbie. > > > > cheers, > > Woody > > > > > > > > Best regards, > > > Dan > > > > > > - Original Message - > > > From: "Mathew Frank" <[EMAIL PROTECTED]> > > > To: <[EMAIL PROTECTED]> > > > Sent: Tuesday, June 03, 2003 10:47 AM > > > Subject: Re: [Asterisk-Users] Voice Modem + Soundcard Driver > > > > > > > > > > > The problem with using Voice Modems is that they fall into two > > > categories: > > > > > > > > > > 1) Hardware Modems which only have half-duplex transmission of voice > > > > > 2) Soft/Win/Lin modems which are proprietry and don't have asterisk > > > > drivers > > > > > > > > or 3 - full duplex "real" voice modems such as produced by Banksia in > > > Sydney > > > > which have been available for years for IVR systems for which only a > > > > single-duplex channel is available in Asterisk. > > > > > > > > > 2 Phone lines > > > > > 2 Banksia Wave SP 336 Modems - these have 3.5mm female jacks for > > > > speaker/microphone > > > > > 1 Full Duplex Soundcard > > > > > 4 3.5mm mono male-male audio cables > > > > > 2 3.5mm stereo male -> 3.5mm female left + 3.5mm female right > adapters > > > > > 1 Linux Box with 2 serial ports and spare PCI slot for Full Duplex > sound > > > > card. > > > > > Asterisk > > > > > Programmer > > > > > > > > I had concidered such an approach, till I remembered the iNTRA modems > > > > mentioned above. There is actually a question re legality of wiring > the > > > > sound card to the modem - or even audio-coupling to do the same > according > > > to > > > > the wording of the rules AFAIK. Anyway I did concider your > suggestion > > > and > > > > when I posted it a year ago got in an argument with a fellow poster > who > > > > missed my point entirely till the point was lost. > > > > > > > > Mathew > ... > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Modem => Serial ?
On Wed, Jun 04, 2003 at 09:05:54AM +0300, Dan wrote: > Hi, > > Serial voice modems use separate jacks for audio in and audio out. > The audio stream cannot be passed through the serial line. Actually Dan, you are mistaken here, many serial fax/data/voice modems come with answering machine programs which tell the modem to send the voice data through the serial line. Also the vgetty program in linux does the same. cheers, Woody > > BR, > Dan > > - Original Message - > From: "Gary" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Wednesday, June 04, 2003 7:17 AM > Subject: [Asterisk-Users] Modem => Serial ? > > > > Now with the usual discussion which arises with compatible internal > > modems for FULL DUPLEX voice etc... > > > > It really makes me think (again) about serial communications > > > > Most external modems will now talk at 115k2 so maybe some will explain > > when/what/how/etc of why a serial interface could NOT be developed ?? > > > > Gary > > . > > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice Modem + Soundcard Driver
On Tue, Jun 03, 2003 at 11:09:28AM +0300, Dan wrote: > Hi, > > I have an USB voice modem which does not need to be connected to the > soundcard in order to build an answering machine. > It appear in the system as another sound card. > I have tested a free Answering machine application and it works great. > The modem is very cheap (paid around 25 USD for it) and has linux drivers, > including the source files: > http://www.smlink.com/products/SmartUSB56.htm for description > and > http://www.smlink.com/download/Linux/slmdm-2.7.14.tar.gz > for the linux driver, including source files. > > Can this be used as a FXO interface for Asterisk? The key is can the modem do full-duplex voice (sound in both directions at the same time). An answering machine is half-duplex: first, the sound is sent from the machine to the caller, then, the sound is sent from the caller to the machine. Also, I don't think Asterisk has drivers for this sort of thing, as most of the hardware is half-duplex. But I am only a newbie. cheers, Woody > > Best regards, > Dan > > - Original Message - > From: "Mathew Frank" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Tuesday, June 03, 2003 10:47 AM > Subject: Re: [Asterisk-Users] Voice Modem + Soundcard Driver > > > > > The problem with using Voice Modems is that they fall into two > categories: > > > > > > 1) Hardware Modems which only have half-duplex transmission of voice > > > 2) Soft/Win/Lin modems which are proprietry and don't have asterisk > > drivers > > > > or 3 - full duplex "real" voice modems such as produced by Banksia in > Sydney > > which have been available for years for IVR systems for which only a > > single-duplex channel is available in Asterisk. > > > > > 2 Phone lines > > > 2 Banksia Wave SP 336 Modems - these have 3.5mm female jacks for > > speaker/microphone > > > 1 Full Duplex Soundcard > > > 4 3.5mm mono male-male audio cables > > > 2 3.5mm stereo male -> 3.5mm female left + 3.5mm female right adapters > > > 1 Linux Box with 2 serial ports and spare PCI slot for Full Duplex sound > > card. > > > Asterisk > > > Programmer > > > > I had concidered such an approach, till I remembered the iNTRA modems > > mentioned above. There is actually a question re legality of wiring the > > sound card to the modem - or even audio-coupling to do the same according > to > > the wording of the rules AFAIK. Anyway I did concider your suggestion > and > > when I posted it a year ago got in an argument with a fellow poster who > > missed my point entirely till the point was lost. > > > > Mathew > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice Modem + Soundcard Driver
On Tue, Jun 03, 2003 at 05:47:54PM +1000, Mathew Frank wrote: > Woody wrote: > > The problem with using Voice Modems is that they fall into two categories: > > > > 1) Hardware Modems which only have half-duplex transmission of voice > > 2) Soft/Win/Lin modems which are proprietry and don't have asterisk > drivers > > or 3 - full duplex "real" voice modems such as produced by Banksia in Sydney > which have been available for years for IVR systems for which only a > single-duplex channel is available in Asterisk. Would it be hard to code a full duplex asterisk driver? Or is it a kernel driver issue? Oh, I see that you can download linux drivers from Banksia... > > 2 Phone lines > > 2 Banksia Wave SP 336 Modems - these have 3.5mm female jacks for > speaker/microphone > > 1 Full Duplex Soundcard > > 4 3.5mm mono male-male audio cables > > 2 3.5mm stereo male -> 3.5mm female left + 3.5mm female right adapters > > 1 Linux Box with 2 serial ports and spare PCI slot for Full Duplex sound > card. > > Asterisk > > Programmer > > I had concidered such an approach, till I remembered the iNTRA modems > mentioned above. There is actually a question re legality of wiring the > sound card to the modem - or even audio-coupling to do the same according to > the wording of the rules AFAIK. Anyway I did concider your suggestion and Austel rules? Seems pretty harsh since Banksia gave me a cable in the box with my modem. It's not like I'm breaking open an austel-approved device and making hardware modifications which would be understandably against the rules. Although from what I hear the Austel rules seem to be about "Systems" rather than "Hardware". Probably linux drivers for winmodems are not Austel approved. > when I posted it a year ago got in an argument with a fellow poster who > missed my point entirely till the point was lost. Are there any Australian FXO Asterisk users? What do they use? Where do they buy their hardware? cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice Modem + Soundcard Driver
The problem with using Voice Modems is that they fall into two categories: 1) Hardware Modems which only have half-duplex transmission of voice 2) Soft/Win/Lin modems which are proprietry and don't have asterisk drivers Please shoot down this recipe before I waste any time trying to acheive it: Rationalisation: In Australia and I assume other places, there is no asterisk-compatible low-end hardware which is legal to use. This could be a way to use existing certified hardware with asterisk. Ingredients: 2 Phone lines 2 Banksia Wave SP 336 Modems - these have 3.5mm female jacks for speaker/microphone 1 Full Duplex Soundcard 4 3.5mm mono male-male audio cables 2 3.5mm stereo male -> 3.5mm female left + 3.5mm female right adapters 1 Linux Box with 2 serial ports and spare PCI slot for Full Duplex sound card. Asterisk Programmer Method: --- Plug modems into phone lines and serial ports as per normal Plug adapters into lineout and linein of sound card Plug modem 1 speaker into left of soundcard linein adapter Plug modem 2 speaker into right of soundcard linein adapter Plug modem 1 mic into left of soundcard lineout adapter Plug modem 2 mic into right of soundcard lineout adapter Adapt soundcard driver and/or other drivers to: initialise modems to use Caller-ID via the serial lines expect RING and caller_id info from the serial lines send AT commands to answer calls, end calls, originate calls get voice data for lines 1&2 from left & right channels of soundcard line in send voice data for lines 1&2 to left & right channels of soundcard line out Diagram: POTS-Line 2 | | Line 1 | | | Line Line - -- |=mic-\ /--mic=| | Modem1 || | | Modem2 | |=spkr--\ /-|---|-spkr=| | | | | | | | | L---R L---R | | | | \ / \ / \ / \ / | | |---Line In---Line Out---| || | Sound Card | || Am I out of my tree? Comments, flames? cheers, -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
