Re: [Asterisk-Users] Testing null values: ast_yyerror(): syntax error
Dnia środa, 18 sierpnia 2004 16:56, Walt Reed napisał:
> OK, I'm going nuts here trying to correctly identify null values,
> specifically when callerID info is not available.
>
> FYI, I'm running Asterisk CVS-HEAD-08/17/04-13:08:53, and Bison 1.875a
> (debian Sid).
>
> A snippit of my dialplan looks like this:
> exten => s,1,SetCIDNum(${CALLERIDNUM})
> exten => s,2,NoOp,${CALLERID}
> exten => s,3,DBGet(temp=idiot/${CALLERIDNUM}) ; Is the person calling an
> idiot? exten => s,4,Goto(s,2001) ; Yep, he's an idiot.
>
Don't know whether there's a more elegant solution, but maybe try putting
values like x${CALLERIDNUM} into the DB? that way if you get a null CID, it
should match against "x".
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Re: [Asterisk-Users] IAX2 (was: diax softphone )
On Wed, 4 Feb 2004 11:13:52 +0200, Dan wrote > Hi, > > From: "Peer Oliver schmidt" <[EMAIL PROTECTED]> > > > >>I have 4569 opened and forwarded/NATed to my *. I am on the same network > > >>as the * server, a friend is remote. After about a minute you loose the > > >>connection. > > > > > This is another problem and it happens for me too (known bug, which > seems to > > > be related with the re-registration which occurs at 60s). > > > > Maybe those two problems are related to each other, i.e. IAX2 tries to > > bridge the call ... (I have no idea what I am talking about) > > > > I think that the call is allways passed through the * server > if IAX(2) is used. > > BR, > Dan > Actually, IAX(2) calls are bridged whenever they can. The server sends TXREQ to both legs of the call and the clients try to connect to each other. What came to my mind is that it might break in NAT environment... Let's see (Note: not tested in the wild, just speculations): - UA1 10.0.0.10:4569 - inside NAT - UA2 1.1.1.1:4569 - outside NAT - server 10.0.0.1:4569/2.2.2.2:4569 - accessible from both UAs UA1 and UA2 register on the server, which fills their apparent_address info the way it can see them (UA1 has private IP, UA2 is public). When a call is established and UA1 and UA2 send TXCNTs to each other, they get the peer address from the * server, so - UA1 tries to connect to 1.1.1.1:4569 which should work (NATted by the router) - UA2 tries to connect to 10.0.0.10:4569 - no go. - UA1's packet reaches UA2 (say, NATted to 2.2.2.2:5), so UA2 sends TXACC, but (correct me if I'm wrong here) the peer address is not updated. Thus, the connection breaks as UA1 never receives the TXACC that UA2 sent. If libiax2 (and chan_iax2 - though I haven't actually looked into it too much) set the peer's address upon receipt of TXCNT, it might work (UA2 would talk to 2.2.2.2:5 which the router would NAT back to 10.0.0.10:4569). If that's not the problem, sorry for confusing you. :) A packet dump should reveal all. What do you think? Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialling Hook Flash on Zaptel
On Tue, 3 Feb 2004 11:33:24 -0600, David Gomillion wrote
> Steven Critchfield wrote:
> > On Tue, 2004-02-03 at 09:53, David Gomillion wrote:
> >> Steven Critchfield wrote:
> >>> On Tue, 2004-02-03 at 06:59, Christopher Lee wrote:
[flames, non-flames etc. snipped]
> what about something like this? NOTE: THIS IS NOT WORKING
> CODE. It is an idea, a concept. If you want to try it to
> make it work, then you will have to build on this.
>
> exten => _*XX,1,Dial(Zap/1/1) (dials a 1 on the outgoing
> zap interface, probably needs a short timeout)
> exten => _*XX,2,Flash()
> exten => _*XX,3,Dial(Zap/1/${EXTEN})
>
> The flash is probably on the wrong side, as I look at it
> more closely. This will probably send the flash to your
> VoIP client. But maybe you could look into scripting with AGI.
>
I'm not sure it'll work as Dial doesn't return to the dialplan. It
seems like the only clean method to do this would be to extend
chan_zap to include flash as an extra digit (a'la W). Would a
copy-paste from app_flash do? I have no zaptel devices to begin with
so I can't really check.
Or maybe some voodoo with transfers, local channels etc. would work?
>
> And with that, I bid the fair Asterisk-User's list a
> farewell, at least for posting. I will now become one of
> the countless other leaches who give nothing back to the
> community. It was good to get help, and I tried to help
> others out, but I have a lot better things to do than spend
> my time helping others only to get flamed every time I turn around.
>
> You need to remember that we're all volunteers. I will only
> take it in the teeth so many times before I say goodbye.
>
> Go ahead and rip me a new one. Have fun. Rant, rave, call
> me stupid. Tell me I have no value, and that I contribute
> nothing. The more you say it, the more accurate it becomes.
>
Hey, why take things personally? I think Steven is known for his
inflammatory posts. Sure it isn't nice to be flamed openly and IMHO
Steven is doing a fine job scaring people away from the list
(sometimes just *too* fine), but we (=the newbies, like me) all end up
learning something. In this way I find his posts doing more good than
harm. From a third-person perspective (not being flamed by him - yet?)
I'd describe his posts somewhere between informative and funny with an
occasional flamebait (yes, I do hang out on /.)
That being said, I'm mostly a lurker myself.
Regards and don't feel too offended :)
Greg
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Re: [Asterisk-Users] Dial via sip gateway?
On Sun, 1 Feb 2004 08:21:55 -0600, Rich Adamson wrote [long snip] > No, the manual is very verbose but no * examples at all. The > box sells as either a 323 or sip, with different images > (sort of like C7960's) and different manuals. > > The box does not support the "register" function in either > direction. I just tried the * sip register, and got a "501 > Not Implemented" with sniffer. > > From what I can tell (box is about 48 hrs old for me), it > seems to be a rather incomplete or just-bare-sip-minimum > functionality. It also appears as though all four ports are > treated as a group-of-lines, and one doesn't have any choice > (from a sip perspective) on which port to use for outgoing > calls. Since this one is set up with 1:home, 2:business, > 3:outgoing calls I really need to be able to select which > port * is going to use, particularly since outgoing 'home' > long distance calls must use a different port then for > outgoing 'business' calls. I have an idea of a crude hack that just might work - e.g. if you need to dial a number on line 3, first make two outgoing calls to a bogus number (just to keep the lines busy for a second) and then place the 3rd call to the destination you want - if I understand the situation correctly, the 1204 should dial on the 3rd line then and the first two calls should drop quickly (no such number). Of course, in that case you need to keep the line state e.g. in the DB so that, say, line 1 in use doesn't mess things up. Yes, I know it's ugly. If it's also bound not to work, I'm all ears as to *why* :) > > The entire box (4 ports) has only a single IP, so if the > dial sip command doesn't have any additional > parameters/strings to destinguish selected ports, guess I'll > return it to the reseller. There appears to be a way to set certain > types of filters on a per port basis in the box, but I can't > see how that could be used to differentiate home vs business > calls, etc. > > Since I don't know anything about 323, does that control > protocol allow some sort of sub-selection where each port > would be addressable? If not, it certainly seems as though > Mediatrix needs to go back to work on their code or something. > > Can you think of any other way that * might interact with > this thing via sip? > > Rich > Regards, Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SayDigits
On Sat, 24 Jan 2004 10:56:59 -0800, Chris Wilson wrote
> Has anyone had this problem:
>
> (When calling to ext. 1010)
>
> Jan 24 10:50:27 WARNING[-1252262992]: file.c:446
> ast_openstream: File digits/" does not exist in any format
> Jan 24 10:50:27 WARNING[-1252262992]: file.c:734
> ast_streamfile: Unable to open digits/" (format ULAW): No
> such file or directory
>
> << in Extensions.conf >>
> exten => 1010,1,SayDigits(${CALLERID})
>
> /var/lib/asterisk/sounds/digits exists, and there are many
> files in there. Any idea's?
>
> Thanks! :)
>
> Chris
have you tried:
exten => 1010,1,SayDigits(${CALLERIDNUM})
?
hth,
greg
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Re: [Asterisk-Users] restricting one user per account
On Thu, 11 Dec 2003 23:46:46 -0800, andrewg wrote > Implementation wise, it would be more "frustrating" to kick > the already registered user off, and make it more likely > it'd be noticed if there where two registered people. > Hmm, frustrating but maybe useful, if you were for example on a dynamic IP box which has less-than-five-nines uptime and your connection dropped. When you get back online, your address is changed and to the * server you're registering to you're somebody different. If *new* registrations were forbidden, you'd have to wait until the old one times out. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Asterisk servers sharing/propagating registry ?
On Tue, 9 Dec 2003 08:48:43 +0100, Nicolas Bougues wrote
> On Tue, Dec 09, 2003 at 08:28:27AM +0100, Florian Overkamp wrote:
> >
> > Registration cascading is not possible (I think) but could it be
solved
> > with a shared dial route:
> >
> > Instead of DIAL(IAX/sip.isp.com) could you not
> > DIAL(IAX/sip1.isp.com&IAX/sip2.isp.com&IAX/sip3.isp.com) to reach
a similar
> > effect ? (or chain them in different lines so it tries to reach
the first
> > one, then the second one if it fails, and the third if that fails.
> >
> > Florian
> >
>
> This could solve the problem in the short term, but is not really
> "elegant". Each time I add a server I'd have to modify quite
> a lot of extensions.conf on each server.
>
hmm how about define
MYSERVERS=IAX/sip1.isp.com&IAX/sip2.isp.com&IAX/sip3.isp.com and use
${MYSERVERS} wherever appropriate? only one change required then
> I'm not quite familiar with IAX(2) and registration
> questions. Does anybody sees any tricky problem that could
> arise with some kind of auto-cascading registrations through
> IAX (that is, any registration, SIP, IAX... would be
> forwarded on an IAX channel) ?
>
> --
> Nicolas Bougues
> Axialys Interactive
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Re: [Asterisk-Users] console sound
On Sat, 06 Dec 2003 10:10:21 -0700, Michael Welter wrote > Hi, > > I have a RH9 system with an onboard VIA sound chip. > According to the archives, VIA won't work for asterisk. > > So, I disabled the VIA and I purchased a Creative Labs > "Soundblaster PCI 128-Voice" soundcard ($13). This card is > on the "approved" RedHat list. However, the documentation > inside the package says "VIBRA 128". > > Anyhow, kudzu doesn't see then card. The soundcard > detection program says "Ensoniq, ES1371 (AudioPCI-97), > module es1371", which seems normal. However, there is no sound. > > Does anyone have experience with this? > > Thanks, > Mike Hi My SB128PCI CT5880 (don't know what's the name on the box, got a raw card w/o anything) half-works on OSS drivers (es1371). I cannot record audio, playback works OK. Alsa (http://www.alsa-project.org/) works fine from command line and from my programs, gnophone bitches about "No input space". After removing the check, works great. Disclaimer: didn't try with chan_oss BTW, with ALSA, you might be even able to use the VIA? my 0.02PLN Grzegorz Nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Soundblaster
I have a SB128 (pci) and via82c686 (on board). I simply did not compile in the drivers for both into the kernel and added es1371 to /etc/modules. adding via82cxxx_audio below would probably give me a /dev/dsp1 or sth :) a long time ago I had two sound blasters (awe64 & pro iirc) and they worked great together. actually now I'm using alsa (snd-ens1371 and snd-pcm-oss in /etc/modules) due to some weird behaviour under oss. hth, grzegorz nosek On Thu, 4 Dec 2003 08:44:06 -0400, Carling R. Messina wrote > Download the latest alsa drivers from sourceforge and make install. > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Michael > Welter > Sent: Wednesday, December 03, 2003 10:27 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Soundblaster > > Hi, > > I have the VIA chipset, and I'm trying to disable the sound > and enable a soundblaster compatible card. > > Can you tell me what you did in /etc/modules.conf to enable your > soundblaster card? > > Thanks, > Mike > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iax Client Library Issues? (DIAX, iaxComm, etc.)
On Wed, 3 Dec 2003 20:45:21 +0200, Dan wrote > Hi, > > - Original Message - > From: "Grzegorz Nosek" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Wednesday, December 03, 2003 5:08 PM > Subject: Re: [Asterisk-Users] Iax Client Library Issues? > (DIAX, iaxComm, etc.) > > > On Tue, 2 Dec 2003 21:11:31 -0500 (EST), firedude wrote > > > The new versions of iaxcomm and DIAX are both now using the > > > iax2 protocol. So in order to receive incoming calls on > > > either of them in your extensions.conf file change > > > IAX/clientname to IAX2clientname. Then you should be able > > > to receive incoming calls on either iaxcomm or DIAX. Also > > > there is a mailing list for the iaxclient library. It's > > > [EMAIL PROTECTED] Hope this helps. AJ > > > > > > > or: > > > > exten=>1500,1,Dial(IAX/client&IAX2/client,30) > > > > This is the best option in the mean time, till the IAX2 > library bug will be solved. DIAX can use both IAX and IAX2, > so please check if with IAX this is not an issue and send > me your feedback. > > Thank you and best regards, > Dan > is this the bug that you mean? filed it today, patch included, works for me (tm). even if it isn't, take a look, it was a big showstopper for me as it essentially blocked any iax2 <-> iax2 call if any client used libiax2 (asterisk itself doesn't). http://bugs.digium.com/bug_view_page.php?bug_id=621 regards, greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iax Client Library Issues? (DIAX, iaxComm, etc.)
On Tue, 2 Dec 2003 21:11:31 -0500 (EST), firedude wrote > The new versions of iaxcomm and DIAX are both now using the > iax2 protocol. So in order to receive incoming calls on > either of them in your extensions.conf file change > IAX/clientname to IAX2clientname. Then you should be able > to receive incoming calls on either iaxcomm or DIAX. Also > there is a mailing list for the iaxclient library. It's > [EMAIL PROTECTED] Hope this helps. AJ > or: exten=>1500,1,Dial(IAX/client&IAX2/client,30) my 0.02pln grzegorz nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
On Mon, 01 Dec 2003 15:14:06 +0100, Michael Bielicki wrote > Low, Adam wrote: > > >Second that ! > > > >-Original Message- > >From: Cees de Groot [mailto:[EMAIL PROTECTED] > >Sent: Monday, December 01, 2003 2:35 PM > >To: [EMAIL PROTECTED] > >Subject: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in > >Paris > > > > > >zoa <[EMAIL PROTECTED]> said: > > > > > >>And while you are in Europe, why not also do Brussels ? ;) > >> > >> > >> > >Amsterdam!! > > > > > > > Warsaw !! :) > Cracow!! :P > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3x AVM Fritz!Card PCI for a EuroISDN PBX.
On Fri, 28 Nov 2003 18:15:38 +0100, Peer Oliver schmidt wrote > David M. Wilson wrote: > > > Hi there! > > > > I'm currently considering various PBX solutions for our office telephone > > network, and would very much like to use Asterisk. Currently, my > > research is incomplete. I have been recommended to use the above cards, > > but it is unclear from my Googling whether my configuration will work: > > > >- 3x Fritz!Card PCI's in one host. > > As far as I know, AVM only allows a single Fritz!Card PCI in > a PC. I /think/ there is a patch out there to allow more > than one. Search the archives to find out more. > > I am sure, you will get better results by putting in an > active card. Either AVM or EICON. I have /heard/ the EICON > cards are preferable because of the on board echo cancellation > -- > Best regards > > Peer Oliver Schmidt > the internet company I'm using two Fritz!Cards in one box without any problems (yes, a patch was required iirc but I found it quite easily via google... somewhere on isdn4linux.de methinks). It works as a router/nat, iax->pstn gateway, mail server, nfs (/home) and smb file server and php+mysql app server for about a dozen clients. It worked for quite a time on a p2/400 but we're currently moving it to a celeron/1700 as the db is getting bigger. No asterisk related problems whatever, except for dead channels left sometimes (to clear them reliably, we need to shut down * and restart it). My 0.02PLN Grzegorz Nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Picking an open channel (FXO port) for outbound calls
On Mon, 24 Nov 2003 11:45:17 -0600, Tony Kava wrote
> Greetings:
>
> I did some quick searching of my history of this list, and I
> tried a quick Google search as well, but perhaps someone on
> the list can quickly answer this question. I have a very
> nicely working Asterisk system at home with two Digium X100P
> FXO cards. When my SIP phones want to dial-out I have them
> setup to grab the first analog card (Zap/1) with the
> following extensions.conf segment:
>
> snippet
>
> ; Outbound
> exten => _9.,1,Dial(Zap/1/${EXTEN:1},90,Tt)
> exten => _9.,2,Macro(fastbusy)
> exten => _9.,102,Macro(fastbusy)
>
> /snippet
>
> Zap/1 and Zap/2 are analog phone lines. What is the best
> method of picking an open line when someone tries to dial-
> out? i.e. if Zap/1 is in use how can I instruct Asterisk to
> use Zap/2 and vice versa? I know complex methods of making
> this happen, but I'm sure there is a very simple way to accomplish
> this task.
>
> Thanks.
>
Try putting the channels in a group (IIRC just put a group=>1 in
zapata.conf before channel=> lines - I don't own any Digium hardware
though :( ) and replace Zap/1/ with Zap/g1/
Works for me with 2x Fritz PCI cards (i4l, as it mostly works and if
it ain't broken, don't fix it.. capi is on my schedule though..)
HTH,
Grzegorz Nosek
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Re: [Asterisk-Users] tunnel iax via gnophone with ssh?
On Thu, 20 Nov 2003 08:44:10 -0700, Chris Hirsch wrote > Hey all...I'm trying to use gnophone to connect to my > asterisk box behind my firewall..I thought I could just > setup a tunnel with something like ssh host.com - > L5036:asteriskserver:5036 and just change my gnophone to > connect to localhost:5036 but I never see anything happen on > the asterisk server. I'm even trying this on the same > network just in case there is something funky with NAT. As somebody already pointed out (and I corrected this in the Wiki a few days ago), IAX uses UDP, so plain ssh tunneling won't work. > > Anybody have any ideas? I did notice that when I start > gnophone I see > > iax.c line 654 in iax_init: Started on port 5036 > Listening on port 5036 > > and it doesn't seem to matter what I do inside the config. > Are these ports in some way hardcoded? If if they are can't > I do something like above? > > Thanks! > Chris The 5036 port is hardcoded in the IAX library (iax.h, #define IAX_DEFAULT_PORTNO 5036) which gnophone uses. For the quickest hack, change the value and recompile libiax. In the longer run, consider adding a command line option (or even better, a GUI config item), specifying an int passed to iax_init() in pc_init() in phonecore.c in gnophone source (I went through the code so much I almost know it by heart ;). Currently it is passed a zero which means "the default". In any case, if the required port is in use, IAX uses a random one. Grzegorz Nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Asterisk Lists (was Re: [Asterisk-Users] Asterisk Business discussion again)
On Thu, 20 Nov 2003 14:54:15 -, Linus Surguy wrote > > So far it seems like the proposed candidates for new lists are: > > > > asterisk-newbies (perhaps a better word?) > > Maybe asterisk-install ? > asterisk-starters ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAQ, Documentation, How-to, etc
On Thu, 20 Nov 2003 00:47:08 -0500, Dorian Gray wrote > I yammered: > > of public resources such as this list. put that FAQ in the list > > subscribe welcome message or the list sig or the asterisk README or > > handbook or all of the above... > > er, in case it wasn't obvious: s/that FAQ/a link to that FAQ/ > I am all for svng prcs bndwdth. Actually a full FAQ sent to all newcomers to the list would be quite useful, I think. As in: I subscribe to the mailing list and the first message I get is the list FAQ (but no-one else sees it, naturally), together with a link to the Wiki, digium's documentation site etc. Alternatively, reading the FAQ might be obligatory to subscribe (must click though it to actually subscribe). my 0.02PLN ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] recording files for menues
On Sun, 2 Nov 2003 16:21:48 +0300, Shoval Tom wrote > Either it's not working, or I don't know what I'm doing. > It's giving me the error "sox: effect '.gsm' is no known! > > Let's say I need to convert file 1.wav to 1.gsm. > How do I apply this command to it? > [snip] > #!/bin/sh > for i in *.wav; do sox $i -r 8000 `basename $i *.wav`.gsm > resample -ql; done > try removing the second *, as in: for i in *.wav; do sox $i -r 8000 `basename $i .wav`.gsm resample -ql; done for a single wav file, try sox 1.wav -r 8000 1.gsm resample -ql (mindlessly copying the -ql, i don't really remember what it does ;) hth, grzegorz nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Absolute Minimum Installation Packages
Hello, On Fri, 31 Oct 2003 10:24:32 -0600, David Gomillion wrote > I can understand the size concerns for putting it in an > appliance or what-not. However, my opinion is that, due to > the low cost of hard disk space, it is cheaper for the > company to go out and buy another hard disk to replace the > extra 500 MB they wasted on a sub-optimal installation than > to pay me to try to get the installation as small as possible. you're absolutely right here, for the cost of a 128mb cf card i can get a 40gb hdd, where the space is not a concern > > What are the benefits to a really tiny installation, aside > from possible appliance applications? Moreover, won't you > still need a sizable hard disk for voice prompts, voicemail > messages, sound file to direct people to dial the correct > extension, etc? what i thought about was a closed box with some web interface that could serve as a voip gateway (and possibly as, say, web proxy/cache held on tmpfs?) not being a full blown server (there's a difference between convincing people to put a 30x30 box somewhere and making them put a high-tech server with raid, streamers & whatnots. having the system run from a read-only medium (like a cf card with a tmpfs overlay - see http://translucency.sourceforge.net though haven't tried it yet) removes the need for backups & extended reliability (nothing changes and if the data is somehow lost, restoring it is trivial). furthermore, if the fs is on a solid state device (not a hdd or cd), there are no moving parts (except for a cpu/sys fan), improving hardware robustness and reducing noise level. as for voicemail, etc. you can put another hdd (capacity!) in there just for that or keep it in ram (speed+no moving parts+cheaper than cf and voicemail tends to have rather short life-time - or doesn't it?). if the hdd breaks or you get a blackout, oh well, you lose at most some voicemail. if i could fit an * distro in 20mb (seems reasonable if started from a floppy-distro), it leaves me 100mb for voice prompts, which should be enough. > > Again, I may be WAY off track, but one of the things I > really like about * is that I can update it easily. > Wouldn't you lose some of the beauty by putting it in an appliance? you can build asterisk on another machine and update it via, say, scp to your heart's desire > > Moreover, I HATE Nortel because they have a user-unfriendly > interface, proprietary controls, non-standard connections, > and the like. It seems to me that by appliance-izing we > would be inviting the same abuses that the current systems > enjoy. I could see it becoming an issue of open-source > software on extremely proprietary hardware, meaning the user > can modify their system if they can figure out how to get in > it. what about ssh? the sshd isn't *this* heavy, is it? putting * in a closed box is appliance-izing it [nice word :)] in the eyes of the end-user (clicks here and there w/o all the *.conf voodoo), but leaves full power to the more competent users who can figure their way through ssh and asterisk's conf files > > Of course, all of this is in the assumption that the end- > user wants to own their PBX. I know I do. I think that we > should be focusing on a useful administrative interface, > database-based extension definitions, and other features > that will advance the power, flexibility, and usability of * > instead of shrinking the distro as much as possible. i think we should aim both to scale up (like, 10k+ phone systems running *) and down (home pbx system with a fritz or x100p and zero initial knowledge required). btw, my shrinking of the * distro to a few dozen mb doesn't stand in the way of expanding your server farm, does it? > > What am I missing? I see many people much smarter than I am > excited about this, so I am sure I simply failed to consider > how it will revolutionize everything. Not that it'll revolutionize anything, it's simply opening another (however niche) market for *. > > Awaiting your enlightenment (preferably sans-flame), > David Gomillion > regards, grzegorz nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Absolute Minimum Installation Packages
On Thu, 30 Oct 2003 20:44:27 -0500, Leif Madsen wrote > > BTW, has anyone already tried to produce a KNOPPIX * Customization? > > Wierd that I had actually started to just think about this earlier > today... :) > > Unfortunately this is going to be nothing that I can do to > help at this point.. I am really quite budgeted for time, > and I can barely work on the other things I have somewhat > commited to. > > I'll be so glad when I'm back in school, and hopefully have > some more time to work on this kind of stuff. > > Keep me posted, I have a couple of idea's that this could be > useful for > (if anything, just what the minimum packages are for a RH > install) > > Thanks! > > -- > +--+ > |Leif Madsen - http://www.hacklocalhost.com| > +--+ > |@| leif at hacklocalhost dot com | > | SMS| sms at hacklocalhost dot com | > | FWD| 18924 IAX| 1700-363-0761 | > |iptel| 8972-1969sipph| 1-747-386-1618 | > +--+ > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users hi i've been thinking of a similar solution myself, but not even on a cd-r but on a compact flash card. the idea behind this was to make a plug'n'play voip gateway of dirt-cheap p2-350 slim boxes. they have the nice property of being almost inaudible (only 1 fan inside) so i thought i could eliminate all possible moving parts and make it still quieter (and less fragile) by removing the hdd and plugging a cf instead (a 128mb cf+adapter costs roughly the same as the computer but oh well ;). so far i've managed to fit quite a complete debian system on ~130mb and if i tried somewhat harder (like installing only the rescue disk image and building on that instead of stripping down a normal install), i'm sure i could get it below 100mb or even smaller. another possibility if you come from the redhat background would be to use [ducks] mandrake, i remember it had (around 8.2 at least) a minimal install which amounted to ~60-~70mb (don't know if redhat has such an option too) regards grzegorz nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Clients not connecting
On Wed, 15 Oct 2003 13:49:23 -0500 (CDT), Dave Weis wrote > On Wed, 15 Oct 2003, M.A. Ali wrote: > > I am kind of new to asterisk. Here is a little prolem that I am facing. > > Here is my problem and questions: I am just adding two gnophone users to > > my dialplan, all three systems are within lan. > > 1. in iax.conf: > > [mako] > > type=friend > > auth=pliantext > Was this copied and pasted or mistyped? > > secret=myown > > context=default > > host=dynamic > > permit=0.0.0.0/0.0.0.0 > > dave on a side note, have you tried with auth=md5? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No sound on PSTN --> */PRI
On Fri, 19 Sep 2003 18:10:37 +0100, Scott Stingel wrote > Have you tried starting asterisk with -c? It should > give you some detail as to what is happening with the call. > > Scott M. Stingel > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of > > Thomas Haeger > > Sent: Friday, September 19, 2003 3:40 PM > > To: Asterisk User > > Subject: [Asterisk-Users] No sound on PSTN --> */PRI > > > > > > Hi all, > > > > i tried to make a call from public pstn in our */E100P. > > Config is following: > > > > exten => _X.,1,Playback(testgsm) > > But what i hear is one dtmf tone and then nothing... > > > > Any ideas ? > > > > > > Regards, > > > > Thomas. [snip] have you tried answering the channel first? as in: exten=>_X.,1,Answer ; a 1-2 sec wait here maybe? exten=>_X.,2,Playback(testgsm) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Radio for Music on Hold?
On Thu, 18 Sep 2003 13:21:54 -0700, Paul Crick wrote
> > Tell your client that some callers put on hold may
> > know about the above and "radio on hold" would make
> > the company look at best ignorent.
> I read something somewhere.. can't remember where.. some PBX
> buyer's guide maybe? ANYWAY.. point is.. it sounds bad to
> the callers.. and you never know what they're hearing..
> dodgy music, a DJ going off on one, throwing a fit, an
> advert for a competitor or something else inappropriate..
>
> Come on people! Fork out $50 for a discman and another few
> bucks for some royalty free library music and have that on
> hold instead.. You're in control, you know what your callers
> are listening to, and you're also legal :-)
>
> Oh yeah.. we're talking Asterisk.. the physical connection
> to an external source is what sparked this whole thread
> off.. sorry, my bad - I forgot.. ok, forget the discman,
> fork out for the music, rip it to MP3 and use the built in
> MOH solution?
>
> Or.. are we still talking about the MOH being the output of
> the radio station that's actually being called, that's using
> Asterisk as its PBX?
>
Hmm, what do you think about about creating a fake extension (like s,
t, fax etc.) called, say, "hold" that would be called every time moh
is played now?
to get the old behaviour you'd do:
exten=>hold,1,MusicOnHold
(or sth)
and you'd get the required flexibility for just about anything.
examples off the top of my head follow:
* dial a sip extension which streams an .asf using some
proprietary/windows/etc. software
* some agi plays you nice music while mixing in some real time
generated info ("you've been on hold for $time. if you're pissed off
already, dial $phone and complain" ;))
* well, the top of my head seems to end here but i'm sure you'll find
more creative uses :)
cheers,
grzegorz nosek
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Re: [Asterisk-Users] LineJack + Asterisk HELP!
On Wed, 17 Sep 2003 09:06:35 -0300, Bartosz Jozwiak wrote > Hello, > > I have kernel-source-2.4.20.tar.gz > and I untar this on. Should I try it once again with tar.bz2 > ? I am ranning the same kernel for sure. > so where's your kernel source (unpacked)? make a symlink from the directory to /usr/src/linux (ln -sf /usr/src/my-kernel-dir /usr/src/linux) just in case and do a: ls -l /lib/modules/`uname -r`/build make sure it points to your true kernel source. if it doesn't, you're *not* running the kernel you're trying to compile for. if i were you, i'd: * download a fresh vanilla 2.4.22 kernel * untar/bz2 it in /usr/src * make a link from linux-2.4.22 to linux * d/l and install openwall maybe? :) * make menuconfig &c. * install the kernel (remember lilo.conf & lilo if you use it!) * reboot to the new kernel * do whatever you desire w/the driver hth, grzegorz nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] using pci modem cards as fxs/fxo ports in *
On Wed, 17 Sep 2003 21:22:59 +1000, Bryan Nolen wrote > Hi all, > > forgive the question but is it possible to use PCI modem > cards (aka winmodem's) as FXO/FXS ports in * ? what about > external modems like the USR Sportsters? short answer: no disclaimer: somebody with better * experience, please review long answer: there are two drivers for analog modems, chan_modem_aopen and chan_modem_bestdata, which correspond to zyxel style voice commands (AT+VRX &c.) and usr/zoltrix style (AT#VRX &c.). off the top of my head, i can't remember which one is which but you can check it easily in the sources. with the two command sets mostly equivalent, the usr driver is incomplete (i hacked it into supporting rates other than 8khz as my zoltrix supports only 7200/11025 but it only rx'd audio. after a stream of mysterious segfaults i gave up trying to hack it into tx'ing audio too ;). i didn't have the chance to try the zyxel driver so i can't comment. there's some code for audio tx in the zyxel driver, but as i said, my attempts to copy the code to the usr driver failed miserably. also i tried a conexant winmodem (yup, still using dialup at home :(). under win98 at+fclass=? gives me 0,1,8 or the like (0=data, 1=fax class 1, 8=voice) but under linux i only have 0. eh, winmodems ;) *however*, i remember a discussion here about one of the * gurus planning to work on the modem drivers - will the brave individual please stand up? :) i'll happily help as i'd be interested in the driver too - haven't checked the development in the area for quite a time though. if anybody is interested in my patches for supporting audio rates other than 8khz (simplistic but acceptable quality sound resample routines right in the driver - good or bad?), let me know, i'll dig them up. > > Thanks in advance, > Bryan. > > Bryan Nolen > Lead Developer > http://Arc.Net.AU > http://cdonline.com.au hth, grzegorz nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LineJack + Asterisk HELP!
On Tue, 16 Sep 2003 12:53:18 -0300, Bartosz Jozwiak wrote > Hello, > > Thanks very much for help. > To install driver for LineJack I need kernel source. > I have debian, and I installed from apt-get install kernel- > source.2.4.20 but while it make ./configure it still asks me > for the kernel source. What can be wrong ? > > -- Bart > hi why not download plain kernel source? anyway, debian kernel-source packages contain only the kernel in .tar.bz2 format (or was it .tar.gz?), named /usr/src/kernel-source-*. you need to unpack it manually and make a symlink to /usr/src/linux probably. btw, make sure you're running the kernel you're compiling the driver for. hth, grzegorz nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Xlite = no sound
On Tue, 9 Sep 2003 09:15:32 +0100, Skuse, Phil wrote > What's the secret to getting sound through Xlite? The SIP > messages all look OK to me, but the sound isn't coming through. > > It was trying to use GSM, so I searched the archive and tried: > > disallow=gsm > allow=ulaw > > Now it says that it's using ULAW but I still get no sound in > either direction. Have you tried setting Send Internal IP to Yes? It worked for me. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hardware IAX phone (please read and reply!)
Hello all! I've talked recently to the head of R&D dept. of Telkom Telos (www.telos.com.pl) - a big Polish company specialised in making phones. I gave them the idea of creating a cheap (cost-effective) hardware IP phone. The phone we discussed would include hardware support for IAX (though probably SIP/sth. else would be required too if it were to hit the market.. what do you think?) and GSM 06.10. Although they have no previous experience in IP phones, they were quite interested and promised to have a deeper look into the issue. So now for the big part: everybody PLEASE give your suggestions about what the IP phone of your choice should look/work/... like. The main reason we started the talks was the cost of currently available phones (even $70+s&h is a truckload of money for a phone here in Poland) but any and all suggestions are welcome. I'd also love to hear from the more hardware-oriented people - do you have any suggestions about used chips, controllers, codecs, whatever? As I said, although they've been making phones for years, they haven't built an IP phone before so they have to research the possible elements used. Why not make it easier for them? :) With Telos being a specialised factory, there's the benefit that e.g. good-looking cases are no problem at all, and if low price wasn't the goal, touchscreens and all would be an option too - maybe some "deluxe" edition? An alternative design that came up was a bigger (say, 12/24 ports) gateway with some embedded Linux running on an industrial PC (as beefy as circumstances require - any comments?) with plain RJ11 sockets on one side and Ethernet on the other. What do you think about this? Hope to hear from you (a lot! :) Grzegorz Nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] weird error message with zaptel
On 20 Aug 2003 15:50:15 +0300, Johanna Kangas wrote > I understand the problem is in hisax. I am a woman, not a > stupid :) > > So anyone who have had same kind of problem WITH HISAX ? > > -Johanna > I, for one, haven't. However, I recommend moving away the whole offending /lib/modules/2.4.x/ directory and doing: make clean make dep bzImage modules modules_install in your kernel source dir (/usr/src/linux-2.4.x or whatever). Remember to install the kernel image in the right place, rerun lilo (if you use lilo of course) and reboot. and read kernel-howto :) OK, I've had my smart-ass five minutes and now it's time for me to ask a newbie question ;) After modprobe capi modprobe fcpci /proc/capi seems ok (shows one card with fcpci driver - sorry I don't post some real output but I had to revert to i4l to make it work as soon as possible) So far, no error messages of any kind, but chan_capi says that CAPI is not installed. My /etc/asterisk/capi.conf is empty (chan_capi demanded it and I didn't know what to put there ;) ;) Thanks in advance Grzegorz Nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Monitor application temporary hack
On 18 Aug 2003 15:07:12 -0600, Jared Smith wrote > On Mon, 2003-08-18 at 14:59, Brian West wrote: > > Maybe we can pester kram to make that an option. monitor.conf anyone? > > > > bkw > > Well, while we're in the "let's pester Mark" mood... why not > have him fix res_monitor so it writes to just one file! > That would sure make me a lot happier... > > Jared Smith on a somewhat related subject, is there a possibility to eavesdrop on an existing connection? like dialling extension XX to eavesdrop on connection initiated by user YY (the mapping can be static if that would be a problem)? the connections would be between gnophone and pstn (i4l driver) and the eavesdropping client would most probably be x-lite if that matters. i need it as the supervisor needs to control and instruct newbie workers ;) tia grzegorz nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] low-cost * (newbie question)
> > > No analog modems. > > > > > > If your ISDN adapter is supported properly, you can place that straight > > > into a asterisk box. The analog line would need a X100P. > > so to sum things up: one x100p for every analog line and one isdn4linux > > adapter for every isdn line, right? i'd like to test it on an experimental net > > with no isdn, so i'd like to use a fake channel to provide something the > > client can talk to. will a soundcard+gnophone on the * box do? or is there > > another channel type, dedicated to testing, with no hardware required? in what > > ways is functionality of the oss channel limited in comparison to, say, isdn? > > >From information on the list, you will probably want a CAPI ISDN adaptor > not an isdn4linux adapter. why? what is the difference in functionality between the two wrt. *? do you mean active vs passive isdn? if i had the choice i'd go for active isdn (e.g. fax support - at least isdn4linux.de or thereabouts says so) but the difference in price is quite significant and i aim to get the cost as low as possible > > gnophone on your * server will require X on your * server, and that > would not be recomended. On my 1ghz AMD chip the screensaver could cause i know, i would never put x on a production server, that's just for testing > a severe degradation on my VoIP channels. Put gnophone on your other > asterisk boxes and call each other. eh, silly me, i haven't thought of that ;) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] low-cost * (newbie question)
> Subject: Re: [Asterisk-Users] low-cost * (newbie question) > From: Steven Critchfield <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Organization: > Date: 01 Apr 2003 09:59:51 -0600 > Reply-To: [EMAIL PROTECTED] > > On Tue, 2003-04-01 at 09:31, Grzegorz Nosek wrote: [snip] thanks for your reply > > No analog modems. > > If your ISDN adapter is supported properly, you can place that straight > into a asterisk box. The analog line would need a X100P. so to sum things up: one x100p for every analog line and one isdn4linux adapter for every isdn line, right? i'd like to test it on an experimental net with no isdn, so i'd like to use a fake channel to provide something the client can talk to. will a soundcard+gnophone on the * box do? or is there another channel type, dedicated to testing, with no hardware required? in what ways is functionality of the oss channel limited in comparison to, say, isdn? > > gnophone supports dialing from URLs, If your application could generate > web pages to feed URL's the people could click on, then gnophone can > accept them from the external app. Otherwise, you may want to watch the the whole app has a web interface so i'd probably go this way > activity and place sample.call files in the queue as your employee needs > a new call to service. > -- > Steven Critchfield <[EMAIL PROTECTED]> > thanks again, grzegorz nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] low-cost * (newbie question)
hello all i'm interested in setting up a small pbx using asterisk and the primary goal is keeping the cost down. the general layout of the net is as folows: * 4 phone lines (2x isdn+ 2x analog) [or 2x isdn + 1x analog, as one might be put aside for a traditional phone/fax with no fancy stuff] * a server box * several client hosts (all linux with x) currently, the isdn lines are unused (to be utilised shortly), one analog line is used by a phone/fax and the other analog line is plugged into the server to provide in/outbound faxes (using hylafax) and the whole net is still sort of experimental in nature. the functionality goal is to provide every client host with a way to make phone calls to public phone network using headsets. currently, the sort of kludgy way i originally thought of is sticking a modem into each box (there are 3 at the moment) to dial the number plus a standard analog handset w/headset connectors (via a pass-through on the modem) to talk. when i found asterisk, i started to wonder whether i could use it in this scenario. i imagine it so: (i'm new to * and telephony issues) * put the isdn adapters and modems into the server box * connect the headsets to sound cards on client boxes (oss-supported onboard ones) * use gnophone to dial the number (i assume i can feed it the number from an external application) and talk using the headset * the call then gets over voip to the server (fast ethernet, same segment) and from there on via one of the lines; either: - one line is set aside for hylafax, the remaining three are for asterisk's use, how is the traditional phone connected to * box then? - one line is set aside for hylafax, one for the traditional phone, leaving two for * does it look reasonably? what equipment is needed? can i use plain standard analog internal modems? how? we really cannot afford much funky hardware (several x100p's, e100p's or an e400p is sort of out of question). thanks in advance grzegorz nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
