Re: [Asterisk-Users] insmod wcfxo failed ( b8zs, esf, wink startis what I'm trying to do.)
>> [EMAIL PROTECTED]:~# modprobe wcfxo >> /lib/modules/2.4.20/misc/wcfxo.o: init_module: No such device >> /lib/modules/2.4.20/misc/wcfxo.o: Hint: insmod errors can be caused by >> incorrect module parameters, including invalid IO or IRQ parameters. >> You may find more information in syslog or the output from dmesg >> /lib/modules/2.4.20/misc/wcfxo.o: insmod >> /lib/modules/2.4.20/misc/wcfxo.o >> failed >> /lib/modules/2.4.20/misc/wcfxo.o: insmod wcfxo failed There's a couple of possibilties here. 1) You've got a T100P card and you're loading the wrong driver. Use the wct1xxp driver for this card. 2) You've got a T400P card and you're loading the wrong driver. Use the tor2 driver for this card. 3) You've got an X100P board and for some reason it just doesn't want to load the driver right. If you're wanting to put a T1 into the system, though, this is the wrong card. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BSD (WAS: Linux flavor?)
> For the development team to get * (and the zaptel cards) running on BSD > shouldn't take too much effort. Perhaps it's just a matter of finding the > right incentive? My only request would be that it be installed to match > BSD > filesytem standards (everything in /usr/local). One of my next projects is to build a driver for the zaptel subsystem and the X100P for NetBSD (I have a non-asterisk voice project and my preferred platform is NetBSD/alpha or NetBSD/sparc...zaptel on an sbus board, anyone? heh) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Seting up TDM40B
> On Fri, 1 Aug 2003 15:25:49 -0500 > "McAughan, Matt" <[EMAIL PROTECTED]> wrote: > >> Have you setup the zaptel.conf and zapata.conf configuration files for >> >how >> ever many ports you have on the card and then run the ztcfg -vvvc >> >command? >> > > Since the module aren't loaded, config zaptel.conf, zapata.conf and run > ztcfg will not work, right? The modules are loading. Its just that the post-install ztcfg is erroring out. > my conf files: > === > zaptel.conf > === > fxsls=1-4 This is why. FXS cards use FXO signalling & vice versa. change this to fxols=1-4 > loadzone = us > defaultzone=us > > > === > zapata.conf > === > callwaiting=yes > callwaitingcallerid=yes > threewaycalling=yes > transfer=yes > cancallforward=yes > echocancel=yes > echocancelwhenbridged=no > rxgain=0.0 > txgain=0.0 > group=1 > immediate=no > context=pstn > signalling=fxs_ks This also needs to be changed to fxo. You've got a mismatch here as well. zaptel.conf is set to use loop-start signalling & zapata.conf is expecting kewl-start signalling. I'd recommend changing zaptel.conf to fxoks and changing zapata.conf to signalling=fxo_ks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Seting up TDM40B
> == Parsing '/etc/asterisk/zapata.conf': Found > WARNING[1024]: File config.c, Line 537 (cfg_process): parse error: No > category context for line 1 of zapata.conf zapata.conf needs to start with the line [channels] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Some questions about a potential usage scenario for asterisk
> There is the other hurdle of clients with existing PBX systems in place. > I've no idea how we'll cover this scenario as I'm sure most clients will > be > reluctant to replace their existing systems, unless of course asterisk can > be "plugged" into some of these systems?!? Yes, it can. If the PBX in question has provisions for attaching an analogue line or extension, then asterisk can be connected to the PBX. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] newbie question - devices
> RE: [Asterisk-Users] newbie question - devicesHi, > > So let me understand this better. > > Asterisk can use SIP gateways which offer PSTN access. For example > www.iconnecthere.com, can be used? > Is this correct? And if it is, than any incoming calls through that > service, could be redirected by astrisk to its users? Yes, asterisk can talk to several SIP gateways for both incoming and outgoing calls. There are also a few providers in the US that provide native IAX call origination and termination. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bugs.digium.com
Is anyone else having trouble accessing it with something besides IE on a Windows box? Opera on Mac/FreeBSD/Linux just hangs at the login page, IE on Mac and Netscape on Solaris & Linux explode when loading login_page.php. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't compile cdr_mysql
> Could you tell me where mysql/errmsg.h is located on your > distribution? We can update the Makefile to look there for that > header. Can't you use mysql-config to get the include and library paths? Granted, you still need to make sure that mysql-config is in your $PATH, but it keeps you from having to search all over Linus's Creation looking for header files. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Get faxed you faxing faxer!
> > I always have a chuckle when I see this. > > it probably could if someone sorts it out, but its reqally starting to > expect a lot. It'll just take someone with the masochistic tendencies needed to do the realtime DSP code for reception. For transmission, however, things are a bit simpler. There's a chunk of code by the late Tony Fisher that can negotiate and transmit group 3 fax if its run on an SGI Indy's sound system. I may be wrong here, but as far as I see it...it would only take a few steps to adapt this code to give asterisk the ability to send faxes. 1) Obtain the rights to use the code. (seance, anyone? Ouija board?) 2) Rebuild the DSP filter coefficients for an 8Khz sampling instead of the 9.6Khz sampling rate the code sets the Indy's sound hardware to. Not all that hard, since one of the programs in the collection of code is the coefficient generator. 3) Adapt the code to run as an asterisk application. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Oen source IP phone, maybe?
Its another one of my "If I only had time...damn this sleep thing" ideas, but I really wonder how hard/cost effective it would be to build an open source IP phone or phone adapter (ala ATA). In about 20 minutes of mulling and research, I figure you could do it for about $40 in parts plus coding time... 1 DS80C400 Ethernet enabled microcontroller with built in IPV4/V6 stack $10 1 DSP56K hardware DSP (you may even be able to dispense with this if the C400 is fast enough to do the codecing...then you just need a 8 or 16 bit DAC/ADC) $10 Misc parts/case/whatnot $20. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Oen source IP phone, maybe?
On Tue, 19 Aug 2003, Michael Sandee wrote: > I guess you will need some software/mem/cpu/flash too? getting it on a > cicuitboard etc? Software would be opensource...get a couple of people together to write it RAM I missed, I thought the C400 had onboard ram, but it doesn't...so add another $10. CPU is what the DS80C400 is. Its a 8051-based microcontroller with a built in ethernet controller and IPV4/V6 stack. Its fairly easy to write code for. There's also no need for much flash...you can DHCP/TFTP the C400. > > You would be more looking at 200$+ for a full board... the thing is you > need something with drivers, or open standards hardware that you can > write drivers for. I've not seen much available boards with dsp etc... > one at broadcom iirc Looking through the C400's documentation again, they indicate that its got some Particularly Beefy(tm) math ability, which would eliminate the need for an external DSP...just need an 8 bit DAC/ADC. > Case is probably not something you make just like that, you usually get > a design, and people make it... and thats very expensive... but it gets For a phone, you are correct. For an ATA-type thingie...a $10 black plastic box from Radio Shack would do just fine. > > I've been dreaming about this aswell though ;) there are just many hooks > to it.. overcome audio problems, speakerphones, echo, echo cancelling... You don't really need all of that for an ATA-type adapter, which is what I was thinking of. Of course a phone based on the same technology would be much more. Oof. Forgot about DTMF decoding from the phone through the adapter. Shouldn't be that hard... Hmmm. Price may be a bit higher than $40...but I'm positive it would be less than an ATA-186 and you'd have full control over the code. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP dialtone?
> > Oh, and let's not forget that the traditional carriers are > > not ignorant > > of what is happening with VoIP or customer interest. There > > is no doubt > > that they are aware that if they don't find a way to deliver > > this service, > > someone else will. No, if they don't find a way to deliver the service, they'll have taxes/laws/regulations passed that restrain other people from doing it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Provisioning CO lines
> Mike, > I opted for an "integrated T-1" for 1 customer who needed about 12 lines. > I configured it with 12 lines voices and 768k data. Chances are you need > this kind of bandwidth if you need 12 phone lines. Combining it on 1 T-1 > can make it a little more cost effective and of course one of the big > advantages is reliability over dsl. They should be able to provide > equipment that will give you 2 T-1 outputs, one of wich you just go straight > into a T110P. Or you take the T1 directly into the T100P and tell the zaptel drivers to use channels 1-12 as voice channels and the other 12 channels as an HDLC/PPP datalink. *poof* Your asterisk box becomes a PBX and a router. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Qwest CallerID question re:[Asterisk-Users] Caller ID problem
> On my SBC phone, I used to hear a high-pitched chirp before the Call > Waiting beep (much like the first chrip of a V.90 modem negotiation tone) > when someone called in and I was on the line. Does this mean SBC was using > FSK to transmit caller ID on my line? Yup. That's CallerID over Call Waiting via FSK ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 to T1 on asterisk?
> > 1. Will Asterisk route from one T1 to another "perfectly"? That > > is, the bits that arrive on the Portmaster would need to be the > > exact bits sent on the PSTN T1. Seem obvious that this should be > > so. > > As of this weekend it does. Can you DACS with it or is it just a passthrough type thing? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialed Number Identification in analog huntgroup
> Again, not near my asterisk box so I can't check this out, > but is it possible to have the different ports drop into * > in a different context for each line? That way you could > just set up an 's' extension in that context for the > different attendants. Yup. Set up different contexts in zapata.conf and extensions.conf for each line (I'm making a rash assumption you're using a zaptel FXO device). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Modems and Tivos? Oh my!
> On Thu, 2003-09-04 at 17:22, Peter Pauly wrote: >> Does the Digium FXS card support modems (and Tivo devices)? >> If so, to what speed have they been tested? Assuming that you can do native zaptel bridging (Going from an FXS port to an FXO port in the same machine), you should be able to get up to 33.6. No 56k, unfortunately, because of the multiple D/A & A/D conversions. If you're codecing the audio and passing it over IP, you should be able to get 33.6 if you use ulaw (non-compressed) encoding. Any of the compression based codecs will most likely make your modem link up at 9600 or a flaky 14.4. > Why would one use dialup for a TiVo. My TiVo has never touched a > telephone line ever. When I bought it I hacked it to work over my cable > modem link using PPP to my workstation and have never risked loosing the > internal hardware. Or spend a few bucks and get yourself one of the ethernet card kits. Then you don't have to worry about ppp connections and you can drag the video off the unit as well. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MP3 streams for MOH: idea
> > [thread change, different topic] >>is > How about a little tiny program that connects to a remote host, grabs > the contents of an MP3 stream, and pushes it into a FIFO locally? It > would be a raw TCP-to-FIFO stream, so mpg123 would be able to digest > it as if it was a local file. The program would take two arguments: > remote hostname/IP and port, and then the file to which the output > would be sent. I don't know how mpg123 handles blocking... Is there any particular reason (rather than not having time to code one and embed it into *) why we can't have our own in-thread connection to an MP3 stream or file, rather than spawning off a process (fork() is expensive as compared to pthread_create()) of mpg123 to play the stream/file? It seems that this spawning/hoping the process dies cleanly is a thorn in a few people's side. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MP3 streams for MOH: idea
>> allow this to happen. Do you know of any tools that convert ASF to >> mp3? > > mplayer/mencoder understands ASF, mp3 and lots of other formats. > Wont play an ASF stream, though...which is what he's looking for. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MP3 streams for MOH: idea
>> Wont play an ASF stream, though...which is what he's looking for. > > you're sure? > > e.g. > mplayer http://live.atlas.cz/radio1/radio1-32.asx > works fine here. > Well, hell. Make a liar out of me. It wouldn't last time I looked. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as a GW or PBX?
> Hi all, > > I've got myself all confused about the capabilities of *. I somehow > convinced myself (because I see a lot emails flying around about IP > phones) > that Asterisk works as a PBX and trunking gateway, but does not do voice > coding (i.e. TDM in, VoIP out). Does Asterisk work as a VoIP gateway that > regular (non-IP, non-SIP) phones can connect to and establish voice > connections to other non-IP or IP phones? The simple answer. Yes. The complex answer. Asterisk can take calls in via TDM, VoIP (via SIP, H.323, IAX/IAX2, and a few others still in experimental stage) and spit them back out on whatever protocol you want. Quite simply, if it needs to be done in telephony, Asterisk can probably already do it. If not, it can be made to do it. > My understanding is that a network would be connected as so: > PSTN phone <---> [a channel bank or something to put voice calls onto TDM] > <> Asterisk <---> RTP stream > > Is this diagram feasible? In this case, Asterisk would be acting as the > VoIP gateway (as well as have PBX responsibilities). Quite feasable and already in place by many asterisk users. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDMoE and codecs
If I have a system with 1 machine to handle incoming H.323 calls and then multiple machines to distribute them to T1 ports over TDMoE, where does the codec translation take place? Does it take place in the master system or does it take place in each of the slave TDMoE systems? Also, any idea how many concurrent G.729 calls a system like this would handle? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDMoE and codecs
> On Wed, 2003-09-10 at 11:55, James Sharp wrote: >> If I have a system with 1 machine to handle incoming H.323 calls and >> then >> multiple machines to distribute them to T1 ports over TDMoE, where does >> the codec translation take place? Does it take place in the master >> system >> or does it take place in each of the slave TDMoE systems? >> >> Also, any idea how many concurrent G.729 calls a system like this would >> handle? > > Not to make you sound stupid, but TDMoE is Time Division Multiplex over > Ethernet. TDM is the same as what is done on T1/E1 lines. TDM implies > either ulaw or alaw encoding depending on your location. So therefore > the codec translation occurs before the audio makes it to TDMoE. That's what I figured. I just wanted to make sure I had my brain in the right direction. > You probably would rather do IAX2 trunking as it uses only the necessary > bandwidth needed for the in process calls. TDMoE will eat the same > amount of bandwidth no matter what is being transported. Add to this the > fact that just like real T1/E1 circuits, they can't be easily > dynamically added. They have to be provisioned and then deployed. IAX2 > will just scale to the ends of your bandwidth and CPU without additional > setup. So have one machine take the H.323 calls in and trunk them out to each of the slave machines via IAX2 trunking (something I've not used yet). And this will move codec processing out to each of the slave machines? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDMoE and codecs
> If the remote ends can do the codec, then yes. If they can't deal with > the incoming codec, then it will be done at your h323 end point. The > benefit of IAX2 trunking is to cut down on your ethernet load and to > make expanding easier. Not to mention IAX2 is much better tested than > TDMoE. Can the IAX2 trunking do the equivalent of "groups" in Zaptel? What I'm needing to do is take a whole bunch of H.323 incoming calls and spread them out across many T1s (more than what would go into a single machine). I suppose I could do something like this in extensions.conf [incomingh323] exten => s,1,Dial(IAX2/foo:[EMAIL PROTECTED]) exten => s,102,Dial(IAX2/foo:[EMAIL PROTECTED]) exten => s,203,Dial(IAX2/foo:[EMAIL PROTECTED]) etc etc etc And then us Dial(Zap/g1) on each of the machines in the incomingiax context. Then I just have to put a G.729 codec on each of the remote machines to handle the G.729 to TDM codecing. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] # of T400Ps in a machine
Is the max recommended still 2 cards, even in a Quad Xeon with superduperwhizbang Hyperthreading? I'll be running incoming G.729 audio out to TDM. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] # of T400Ps in a machine
So 3 or more TE410Ps in a system? Is the bus mastering design that much of a significant improvement? > I would strongly consider the TE410P in this configuration and would be > interested in working with you to check scalability. > > Mark > > On Wed, 10 Sep 2003, James Sharp wrote: > >> Is the max recommended still 2 cards, even in a Quad Xeon with >> superduperwhizbang Hyperthreading? I'll be running incoming G.729 audio >> out to TDM. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
On Fri, 12 Sep 2003, Jim Paraschou wrote: > I have problem with a TDM40B installation. > When i modprobe wcfxs the error i get is the > following: > > /lib/modules/2.4.19-4GB/misc/wcfxs.o: init_module: No > such device > Hint: insmod errors can be caused by incorrect module > parameters, including invalid IO or IRQ parameters. > You may find more information in syslog or the > output from dmesg > /lib/modules/2.4.19-4GB/misc/wcfxs.o: insmod > /lib/modules/2.4.19-4GB/misc/wcfxs.o failed > /lib/modules/2.4.19-4GB/misc/wcfxs.o: insmod wcfxs > failed > > Does anybody know the poblem? Means the module can't find the card anywhere. Is the card inserted properly? Does it show up if you do an lspci? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Analog FXO Card
>> > Interesting that it has 2 ports on it, and a speaker. The picture looks > a whole lot like a modem to me. The real X100Ps look like a modem too. They have 2 ports and a speaker. When I misplaced mine, I rummaged around looking for it and kept finding it but putting it back in the pile thinking "Nah, that's a modem." ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T/E410P motherboard requirements ?
> Hi, > > Can anyone suggest a good motherboard for the T/E410P cards ? Coz it > doesn't get inserted in the the regular P4 motherboards due to PCI slot > (32 bit) Any suggestions. > I'm an AMD Athlon bigot, I'm using the MSI-6501 dual AMD MB. Its got 2 64-bit PCI slots that'll take a TE410P. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P & T100P knock-off boards
Do they fall under FCC certification if they're built to the same specifications as the ones from Digium? If I build my own T100Ps from the schematics and board layouts that are available, are they legal to plug into the PSTN? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing call spool
On Mon, 29 Sep 2003, Bill Leckey wrote: > I've been playing with the outgoing call spooling feature a bit lately > and it all works as it should with the exception of one irritation. > > I'm mostly using SIP to talk to the phones and using G.723.1 > > I copy the call file into the spool/outgoing directory and the > originating phone rings. I pick it up and the remote phone rings. > However there is dead silence from the originating earpiece. Is it > possible to somehow generate a ring in the earpiece until the remote > phone is picked up? Asterisk can neither originate or terminate calls using G.723. You'll need to use another codec. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nortel M Series phones support
> -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > I've searched the mailing list quite extensively, but didn't come up > with anything promising (some things wer helpful, though). Does anyone > know if Nortel M Series (specifically the 2008, 2616, 7208, and 7310) > phones can be made to work with the TDM400P card or if they are ADSI > compatible at all? I kind of doubt they will work if they are not > compatible, but I don't know what it would take to plug them directly > into a * box. They're proprietary to the Nortel systems. They're fully digital phones, not analog with ADSI ontop. Won't work with *. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF weirdness
I've got a handful of T1s going into a TE410. When I place calls into the system over these T1s, the system either doesn't decode all of the DTMF digits or it decodes ones that aren't there. When the system places calls out, there is no problem doing the DTMF detection. Everything works great. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF weirdness
> On Wed, 2003-10-01 at 16:40, James Sharp wrote: >> I've got a handful of T1s going into a TE410. When I place calls into >> the >> system over these T1s, the system either doesn't decode all of the DTMF >> digits or it decodes ones that aren't there. >> >> When the system places calls out, there is no problem doing the DTMF >> detection. Everything works great. > > answer the line. Tried it with both a specific answer in extensions.conf as well as answer in AGI. still won't decode right. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
> Actually, if this was to be done, it might be an idea to do it with DNS, so > client machines would just do > Dial(IAX2/[EMAIL PROTECTED]/442071234567) and the DNS > system would resolve which machine is the correct target - no cleverness at > all required at the client end, so implementation would be portable across > all the other gnophones etc. Yup. That would be the way to do it. I'll contribute the DNS code for it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
> It is that type of mechanism that enum uses and yes it was to solve a > similar goal, but in this case you need a 'route server' type system - in > particular as this is for IP routing of PSTN end points not on an IP > network. A discussion about this came up a while ago. I suggested something along the lines of BGP, where each endpoint announces "prefixes" of what they can get to. You'll need a central machine that everyone peers up with and then you can use a switch => statement or exten => _.,1,Dial in * to query that machine and get the best route for your call. If you make sure that your destination machines are not behind NAT or a firewall, you can do an IAX handoff to get the connection set peer to peer instead of through the central server. Example: 4 remote * machines, each configured with our "BGP" software. Machine 1 announces that it can terminate calls to country code 1 with a cost of .02. Machine 2 announces that it can terminate calls to 1 with a cost of .05. Machine 3 announces that it can terminate calls to 1-830 with a cost of 0. Machine 4 announces that it can terminate calls to 1-830-751 with a cost of 0. You place a call to 1-830-751-2000 and the system determines that it can place that call for a cost of 0 to machine 4. You place a call to 1-240-988-4000 and the system determines that it can place that call via either machine 1 or 2, but lowest cost is machine 1. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remote control IVR
> Which one would one should I use to solve my problem? Does an loadable > application give you more control than an AGI script? > If you want something that runs continuously (such as a listener process or control process), it'll have to be a loadable module. AGI scripts only get run when the extention they're configured for is called. > > > Best regards, > Ívar Ragnarsson > Grunnur ehf. > Iceland > > ps. Does a loadable application have to be GPL licensed? Only if you plan to distribute it with asterisk, I believe. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system?
On 11/22/2013 12:52 PM, Todd R. wrote: Just checking one more time to see if anyone has an opinion on this. I am primarily interested in using a cloud type setup such as Amazon AWS for the redundancy, easy backup and recovery options. It's not about price but the idea that it will be very hard for a single piece of hardware to ruin my day. I have only one small datapoint. I ran an EC2 microinstance with Asterisk and a dozen offboard users. The only problem I had was SIP wasn't dealing well with the Elastic IP one-to-one NAT that Amazon uses. I had the usual Asterisk/NAT issues of one-way audio. I eventually moved from Amazon to Linode to get away from the NAT issues. Once I did that, everything worked fine, but again it was only a dozen users. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Amazon, Asterisk and reliability beyond a hobby system?
On 11/24/2013 2:47 AM, Todd R. wrote: Did you have the externalip setting in sip.conf set to the Elastic IP? I believe I did. But I didn't really get a chance to plow into it too much, I had a client holding me at gunpoint. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] link to MySQL connection
On 12/3/2013 10:11 AM, Don Kelly wrote: In the php routines, I would like to use the persistent connection that is established in the dialplan, rather than creating a new connection each time they run. How can I do this? You can't, they are completely separate processes and code. Joshua Colp Thanks--that's not the answer I wanted, but it sure was quick. :) Is there anything that would enable me to use a persistent connection in the agi? --Don Yes. Use func_odbc in your PHP AGI. In Asterisk dialplan functions are treated like dialplan variables so you can get and set them just like you would other dialplan variables. If it takes 5 seconds to open a PDO DB connection inside PHP you have some OTHER problem. Eric Wieling Thanks, Eric. If I can't figure out why the connection takes so long, I'll try the func_ODBC approach. look into DNS problems. Something may be trying to do an reverse DNS lookup and taking too long to either give up or get an answer. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk AMI - PHP or Node.js?
On 12/28/2013 7:04 AM, Shahid H wrote: Thanks Daniel, that was useful, I will check those links :) I am pretty good with PHP and jQuery. So I guess learning Node.js shouldn't be too difficult. If I decided to use Node.js - what is the best way to communicate with a browser to AMI process? Send a XML or HTTP command from a browser to AMI process .. or whatever I execute on the browser - it save the commands to the database.. a process will listen the commands from a database. Other options are http requests and do the ipc... or maybe WebSocket? My code will probably end up on Thedailywtf.com for it, but I have a PHP-based system that uses PAMI to talk to the AMI interface. There's a daemon process that listens for commands via a FIFO buffer. Several other processes & web scripts just send commands to the controller process via that FIFO. Seems to work well for me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No compatible codecs, not accepting this offer!
On 1/15/2014 3:59 AM, Francesco Namuri wrote: Hello, I'm having this issue on my pbx, it appears that asterisk is refusing the codecs that my providers is proposing. My trunk configuration is: Pretty simple - --- username=5x5x7x9x0x3 type=friend secret=CRcxn7sqwm qualify=yes port=5060 insecure=port,invite host=sip.txtxlxoxp.it fromuser=5x5x7x9x0x3 fromdomain=sip.txtxlxoxp.it disallow=all context=from-trunk allow=alaw Here you're disallowing all codecs except alaw. --- A typical invite from my provider is: <--- SIP read from UDP:xx.yy.xx.yy:5060 ---> INVITE sip:5x5x7x9x0x3@192.168.1.168:5060 SIP/2.0 Via: SIP/2.0/UDP xx.yy.xx.yy:5060;branch=z9hG4bKt5sfh7nrvok3d5gqc3ritdv7b7 From: ;tag=SDdgce901-90915 To: "SIPLineUser SIPLineUser" Call-ID: SDdgce901-9cb68ba025684f03a4094ed71e6e04f8-ao92gd1 CSeq: 59458 INVITE Content-Type: application/sdp Contact: User-Agent: Nortel SESM 14.1.0.12 Max-Forwards: 19 Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,gin,com.nortelnetworks.im.encryption,replaces,100rel Remote-Party-ID: ;screen=yes;screen-ind=0;party=calling;counter=0;npi=NPI_E164;ton=TON_NATIONAL P-Asserted-Identity: Allow: UPDATE,REFER Content-Length: 293 v=0 o=- 0 138163748 IN IP4 xx.yy.xx.yy s=IMSS e=unkn...@invalid.net c=IN IP4 xx.yy.xx.yy t=0 0 m=audio 43718 RTP/AVP 8 18 3 101 a=fmtp:18 annexb=no a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000/1 Butprovider will only send GSM or G729. So either you need to talk your provider into sending alaw or you need change your allow line to "allow=alaw,gsm". -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No compatible codecs, not accepting this offer!
On 1/15/2014 5:50 AM, Gareth Blades wrote: On 15/01/14 09:39, Francesco Namuri wrote: Hello James, thanks for your answer, I supposed this too, but my provider answered me that as m=audio 43718 RTP/AVP 8 18 3 101 ^ ^ ^ GSM proposal ^ ^--- G729 proposal ^-- aLaw proposal And that a=rtpmap:18 G729/8000 proposed as media conversion a=rtpmap:3 GSM/8000/1 because the call is made by a mobile I would agree with what your service provider has said. If you look at the RFC http://tools.ietf.org/html/rfc4566#section-5.14 the '8 18 3 101' parameters are a list of media formats. The first is the one which should be used but (preferred choice) but the other may be used. Numbers in the range 96-127 are dynamic payload types and these must have a corresponding 'a=' line specifying the payload type and the codec options. Lower numbers have static payload assignments and according to that RFC dont have to have corresponding 'a=' lines. A list of types can be found at http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml However in all SIP traces I have seen there has always been a 'a=' line for every payload type offered. The static payload type numbers are used but there is still the 'a=' line. I missed the RTP/AVP line. I'll go back to lurking for a bit :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] answering machine screening with MixMonitor
On 2/5/2014 12:09 PM, G. Paul Ziemba wrote: I'm using asterisk 1.8 as an answering machine. I'd like to hear the calls it answers aloud in case I want to pick up and interrupt the call. There are a few articles describing, for example, three-way calling a monitor phone set to auto-answer, but I couldn't find anything that described how to just send the audio to a local speaker. A local speaker connected to the Asterisk box itself? Console channel driver, chan_alsa (or chan_oss for old drivers). You'll probably end up with kind of a Rube Goldbergish approach, probably something involving ChanSpy or a conferencebridge to take the place of mixmonitor. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Host = Dynamic in a Register Free Setup
On 2/18/2014 2:09 PM, Eric Wieling wrote: No. Asterisk will accept calls from unregistered devices, but you have to enable guests I sip.conf and hope your dialplan is secure. No sane person does this. Asterisk cannot send calls to a device unless it knows the address from a register or from a host= entry for the peer. You may not like it, but this is the way Asteirsk has worked for the past 15 years. Isn't there also autocreatepeer? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Which is more efficient for 1 to many broadcasting?
Putting a whole bunch of people into a listen-only/muted Confbridge conference or getting the broadcaster audio into a MOH class and then just having callers attach to that MOH class? Does the the muted side of a Confbridge Room still try to mix in audio from the muted channels or does it just disregard those channels and only run mixes against unmuted channels? Now, if the answer is "MOH is more efficient", can someone suggest a way for a channel to be the source of a MOH class? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which is more efficient for 1 to many broadcasting?
On 3/18/2014 6:58 PM, Paul Belanger wrote: On Tue, Mar 18, 2014 at 1:02 PM, James Sharp wrote: Putting a whole bunch of people into a listen-only/muted Confbridge conference or getting the broadcaster audio into a MOH class and then just having callers attach to that MOH class? Does the the muted side of a Confbridge Room still try to mix in audio from the muted channels or does it just disregard those channels and only run mixes against unmuted channels? Now, if the answer is "MOH is more efficient", can someone suggest a way for a channel to be the source of a MOH class? What sort of channel count are you looking for? We did some load testing recently and found less people in a bridge is better then more. Audio source location didn't really matter much. A few hundred to start with, but as with everything, I'd like to scale up as far as I can. And, of course, it makes sense that "less people in a bridge is better than more" but that's not quite what I'm asking. Is it more efficient to have, for example, 701 people in a confbridge room (700 muted users + 1 person yapping) or to have 700 people dialed in and just running the MusicOnHold application with said person yapping away via some audio source. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Numbers hackers call
On 3/26/2014 12:20 PM, Michelle Dupuis wrote: If this is to 972 area code then the next digits should be 0X or 0XX but they are not. This differs from what I found documented for that area code - I thought someone from the region might add to the discussion. Not sure if this reflected a premium service etc. (But someone jumped in with an explanation) 0X or 0XX is only if you're in country and need to dial with the 0 national trunk code (much like dialing 1+ in the US for an in country but long distance call). Someone dialing from outside the country doesn't need to add the zero, so they just use the 972 country code + 59 prefix. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Asterisk Polling Solution
On 4/21/2014 1:47 PM, Mitul Limbani wrote: Use vicidial for achieving the same. Or call files (or AMI originate), a short bit of dialplan logic, and maybe a call to Queue(). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Asterisk Polling Solution
On 4/21/2014 3:58 PM, Nick Cameo wrote: On Mon, Apr 21, 2014 at 2:01 PM, James Sharp mailto:ja...@fivecats.org>> wrote: On 4/21/2014 1:47 PM, Mitul Limbani wrote: Use vicidial for achieving the same. Or call files (or AMI originate), a short bit of dialplan logic, and maybe a call to Queue(). This is a nice and easy solution however, I do not know where to begin. Can you gents kindly elaborate or point us to the right directions (ie, howto tutorials) Asterisk call files: http://www.microalcarria.com/descargas/documentos/Linux/varios/Asterisk/asteriskdocs-docbook/docs-html/x1512.html (Replace Channel with dial commands to your telephony provider, change the context and extension with an appropriate). Dialplan: You'll probably just need some Playback and Read commands to play your message and get the digit response. Agents and Queues: http://www.voip-info.org/wiki/view/Asterisk+call+queues You'll probably want to use static/fixed agents unless you have a whole bunch of agents logging in and out. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Asterisk Polling Solution
On 4/22/2014 5:54 PM, Nick Cameo wrote: Hello Everyone, Thank you all for your response. The people I am doing it for run a non-profit charity, and are legally able to reach out to their customers. I will wire it up to the DNC however, for starters, I would like to get asterisk to: i) Iterate through a list of numbers ii) Play a pre-recorded message asking if they have waste they need picked up iii) If they press one, forward the call to mailbox That's about as simple as it gets. A call file that goes to the dialplan. A dialplan that consists of Read (which would play the message) followed a GotoIf into a mailbox (either voicemail or Dial() to an external number). One hint for doing unattended dialing like this, make sure you're dialing using a SIP or other digital method rather than, say, out an analogue port that doesn't have decent answer detect. And you can't just dump a whole bunch of call files into the system at once, you'll need to meter them out based on the number of concurrent outbound calls your provider will allow. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Open Source Asterisk Polling Solution
On 4/23/2014 12:20 AM, Nick Cameo wrote: That's about as simple as it gets. A call file that goes to the dialplan. A dialplan that consists of Read (which would play the message) followed a GotoIf into a mailbox (either voicemail or Dial() to an external number). One hint for doing unattended dialing like this, make sure you're dialing using a SIP or other digital method rather than, say, out an analogue port that doesn't have decent answer detect. And you can't just dump a whole bunch of call files into the system at once, you'll need to meter them out based on the number of concurrent outbound calls your provider will allow. Hello James, Good to see you here, and thank you very much. Though my basic idea of how it will look using call files and dialplan is like what you and others on here have pointed out. Yes, we are using SIP for both origination and termination (just helping my friend use some of our accounts used for PBX, and prepaid). I have been using * for many years now however, never for call center/predictive dialer type processes. Once I have got this thing to call out and get calls coming in. It would be nice to write to a database all the users that press option on. I have a strong Java, PHP and SQL background. Will probably need to make a call using AGI or such? N. You can go AGI, but there are direct ODBC handles available in the dialplan if you build Asterisk properly with the ODBC resources enabled. That'd my personal preference from a performance standpoint. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] "CBAnn" channel not going away in Asterisk 12
On 5/1/2014 10:38 AM, Richard Kenner wrote: Please go ahead and open an issue and attach the refs log and the full DEBUG log. That will allow us to understand what's occurring here. I need to wait until I'm sure this isn't something I caused somehow, so I need to first understand why I'm seeing this and nobody else is. I had seen it as well but just chalked it up to not grokking how the CBAnn channels worked. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interesting new hack attack
On 5/22/2014 12:41 PM, Steve Murphy wrote: So, these defenses can be employed to stop/ameliorate such hacking efforts: 1. Keep your phones behind a firewall. Travellers, beware! Never leave the default login info of the phone at default! 2. Never use the default provisioning URL for the phone, with it's default URL or password. 3. Use fail2ban, ossec, whatever to stymie any brute force mac address searches. 4. Use your firewalls to restrict IP's that can access web, ftp, etc, for provisioning to just those IP's needed to allow your phones to provision. 5. Keep your logs for a couple years. 6. Change your phone SIP acct passwords now, if you haven't implemented the above precautions yet. If I missed a previous post on this, forgive me. Just thought you-all might appreciate a heads-up. Encrypt your provisioning system if the phone supports it. I had a cable/voip service provider who HTTPS provisioned by MAC without encryption and the provisioning URL was stored, unlocked, in the ATA. Had I been slightly more nefarious, I could have walked the the provisioning tree nice and slow and easily grabbed everyone's SIP credentials in the clear. No hacking or cracking was involved. The ATA doubled as the NAT router they handed out and gave the admin password out freely. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: * with RADIUS
> Can someone give me an idea exactly what things are intended to be tested > via RADIUS, or some other AAA system? > > Are we talking about building SIP/IAX/H323 entries from RADIUS? > This is where the PAM system I developed for * comes into play. I've got most of it working at the moment, but I'm having trouble figuring out how to actually pass the authentication information back to the requesting channel driver. It seems that the structs & linked lists that the drivers use to authenticate are only built on a restart/reload. I'm not sure how to handle this part of it dynamically. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium Wildcard TE410P
It can be either. > Does this card only work as PRI or can it be used like a standard T-1 > wired > to a PSTN Switch? > > TIA > > -Seth > > ___ > 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] FAX, IAX and *....Maybe I'm dreaming...:-)
> It's just my lowly opinion but I too must agree when it comes to the > consumer/soho (1 to 3 line) markets. > > CAUTION!!, DANGER!! Marketing Hat On!! > > Vonage, the most "visible" marketer of a voip consumer product must also > agree. Vontage offers an ip "fax line". using cisco's ata. Vontage must > see > some good reason for doing so. (I assume it's h.323). A&T, MCI and Time > Warner will be competing directly against Vonage when they introduce their > consumer voip products. I'd bet they too will be offering an ip fax line. > > Odds are you will be competing against them too. If I was vonage I'd be > telling the world how important a ip fax line was :-) Personally, I dont think that the world in general really cares about an "ip fax" line. All they want is a system that works all the time/every time and doesn't require elaborate and convoluted setup. They're not ooohing and ahhhing about "oooh, this uses VoIP". They just know that they can stop spending $30/mo on an analog phone line and they get their long distance either flat rate or for an absurdly low per minute rate. Don't sell it as VoIP. Sell it as a total replacement for the analog line. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Headless Linux system for Asterisk
> Run using a serial console > (http://www.tldp.org/HOWTO/Remote-Serial-Console-HOWTO/). No monitor, > VGA adapter, keyboard etc needed. Use SSH to log into the asterisk box > for any maintenance, etc. If the box gets hosed, connect the serial > port to a working PC and fire up minicom and your all set. You'll find > this type of setup quite often in data center environments. Except there is a known problem of dropping/missing interrupts with running serial consoles with certain Digium boards. You also have the same problem if you use a framebuffer console. If you truly want it headless with a serial console without that problem, stick a PC Weasel board in it (http://www.realweasel.com). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Land line vs. VoIP provider.
What about having your VoIP gateway system placing a 911 call to the 911 answering center in the appropriate region and when the 911 operator answers, have a message say "This is a 911 call from 123 Main Street, Nowhere Nebraska" then connect the caller to the 911 operator. Legal? Maybe. Dunno. Just a random thought that I came up with on the way to the aforementioned middle of nowhere, Nebraska. > > Not all VoIP providers will have Vonage's 911 issues. It's perfectly > possible for a VoIP provider to provide outbound caller information to > the PSAPs if they spend the time and money to do so. > > Stephen > > >> Summary: if you're the only caller, calling only to the US, then you >> might be crazy to not use a land line, especially given the deals >> currently available and the 911 issue (but see >> http://www.vonage.com/features_911.php). Even then, if you already > have >> broadband in house (or at home), VoIP amy be an attractive > alternative, >> if only for the control it gives you over your phone service. >> >> ___ >> 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] frame buffering
> > Hi all. > > Could it be possible that video frame buffering be causing problems > even if the computer is not running X ? Yes. There are known problems with systems running with either a frame buffer console or a serial console. For best results, run a plain VGA console. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with x101P
> Occasionally I do NPA-NXX lookups for my local exchanges to see what other > carriers have prefixes in my area. I used to use telcodata.us, but they > seem > to have gone offline. Usually, after you find the carrier's name, you can > see info on the location and type of switch being used. I can't say with > any > assurity that the info is accurate, but it is there if you dig http://www.dslreports.com/coinfo Has the same kinda info. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Web Dialer
> I am putting together a solution that will employ the Digium TE410P with > one T1 going out the PSTN and the other front-ending a PBX. The idea is > that based on a URL, Asterisk will dial an employee behind the PBX. When > the employee picks up, Asterisk will dial the customer (detailed in the > URL). I am assuming Asterisk can work with Apache (through AGI maybe) to > dial the employee and then connect to the customer via info in the URL > (or related through some sort of DB lookup). Another requirement will be > to record the phone call as well. You could do it through either the Asterisk manager interface or have a CGI scrip t in your web front end create an auto call file that dials the employee and runs a second Dial command upon answer. > I have worked a bit with Asterisk and am very happy with what it can do > -- and would prefer to stay with Asterisk. The question is, can Asterisk > handle what my requirements are or would this better be served by > Bayonne? Asterisk is better. Hands down. No questions. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
>> Andrew Kohlsmith wrote: I would set the "Enterprise Class" bar at five 9's reliability (about 5.25 minutes per year of down time) the same as a Class 4/5 phone switch. This would require redundant design considerations in both hardware and software. >>> >> >> To turn around, let's discuss what we need to focus on to get >> Asterisk there: >> >> Here's a few bullet points, there's certainly a lot more >> * Linux platform stability - how? > > Even more than Linux itself is the x86 platform... I've thought about this > a bit when considering * boxes for big customers. When one actually comes > along, I'll have to actually make a decision :-). >>From where I stand, the best thing to do for smaller customers is give > them a box with RAID and redundant power supplies, if they can afford it. You can overcome most of those problems by buying good quality hardware. If you buy your * server from your local Taiwanese clone shop, you're asking for trouble. A big, beefy machine from Dell would be better. > But if I were to have a big customer with deep pockets, I'd really like * > on a big Sun beast with redundant-everything (i.e. you can hot swap any > component and there's usually n+1 of everything). The problem is that I > don't think there's any Solaris support for Digium cards, since it's kind > of a chicken-and-egg problem. Nope. No Solaris support, but you might be able to get away with Linux/Solaris...but then you lose a lot of the hot-swapability. In my experience, though, the only things I've ever been able to hotswap were power supplies and hard drives...and thats not software/os dependant. > One of these days, I may convince myself to buy a modern Sun box (maybe > the ~$1000 Blade 100s) and see what can be done. The only problem I could > conceive would be endian-ness, but I read about Digium cards in a PowerPC > box, so that won't be a problem, right? > Nick Endian-ness is really only a driver issue. Its when programmers who believe that the world revolves around Linux/i386 that you have problems. Personally, I'd stick my Digium cards into an Alpha of some sort. A DS-10L for 1U mounting with 1 card or a DS-20 for multiple cards where you need lots of processor zoobs. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway
> 1. Moving a physical interface (whether a T1, ethernet or 2-wire pstn) is > mostly trevial, however what "signal" is needed to detect a system failure > and move the physical connection to a second machine/interface? (If there > are three systems in a cluster, what signal is needed? If a three-way > switch is reqquired, does someone want to design, build, and sell it to > users? Any need to discuss a four-way switch? Should there be a single > switch that flip-flops all three at the same time (T1, Ethernet, pstn)?) Simple idea: Have a process on each machine pulse a lead-state (something a s simple as DTR out a serial port or a single data line on a parallel port) out to an external box. This box is strictly discrete hardware and built with timeout that is retriggered by the pulse. When the pulse fails to arrive, the box switches the T1 over to the backup system. > > Since protecting calls in progress (under all circumstances and > configurations) is likely the most expensive and most difficult to achive, > we can probably all agree that handling this should be left to some > future long-range plan. Is that acceptable to everyone? Its going to be almost impossible to preserve calls in progress. If you switch a T1 from one machine to the other, there's going to either going to be a lack of sync (ISDN D-channels need to come up, RBS channels need to wink) that's going to result in the loss of the call. > 2. In a hot-spare arrangement (single primary, single running secondary), > what static and/or dynamic information needs to be shared across the > two systems to maintain the best chance of switching to the secondary > system in the shortest period of time, and while minimizing the loss of > business data? (Should this same data be shared across all systems in > a cluster if the cluster consists of two or more machines?) > > 3. If a clustered environment, is clustering based on IP address or MAC > address? >a. If based on an IP address, is a layer-3 box required between * and > sip phones? (If so, how many?) Yes. You'll need something like Linux Virtual Server or an F5 load balancing box to make this happen. You can play silly games with round robin DNS, but it doesn't handle failure well. >b. If based on MAC address, what process moves an active * MAC address > to a another * machine (to maintain connectivity to sip phones)? Something like Ultra Monkey (http://www.ultramonkey.org) >c. Should sessions that rely on a failed machine in a cluster simply > be dropped? >d. Are there any realistic ways to recover RTP sessions in a clustered > environment when a single machine within the cluster fails, and RTP > sessions were flowing through it (canreinvite=no)? >e. Should a sip phone's arp cache timeout be configurable? Shouldn't need to worry about that unless the phone is on the same physical network segment. >f. Which system(s) control the physical switch in #1 above? A voting system...all systems control it. It is up to the switch to decide who isn't working right. >g. Is sharing static/dynamic operational data across some sort of > high-availability hsrp channel acceptable, or, should two or more > database servers be deployed? DB Server clustering is a fairly solid technology these days. Deploy a DB cluster if you want. > 4. If a firewall/nat box is involved, what are the requirements to detect >and handle a failed * machine? >a. Are the requirements different for hot-spare vs clustering? >b. What if the firewall is an inexpensive device (eg, Linksys) with > minimal configuration options? >c. Are the nat requirements within * different for clustering? > > 5. Should sip phones be configurable with a primary and secondary proxy? >a. If the primary proxy fails, what determines when a sip phone fails > over to the secondary proxy? Usually a simple timeout works for this..but if your clustering/hot-spare switch works right...the client should never need to change. >b. After fail over to the secondary, what determines when the sip phone > should switch back to the primary proxy? (Is the primary ready to > handle production calls, or is it back ready for a system admin to > diagnose the original problem in a non-production manner?) Auto switch-back is never a good thing. Once a system is taken out of service by an automated monitoring system, it should be up to human intervention to say that it is ready to go back into service. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Sun Servers with UltraSparc Processors
> I had documented the Makefile modification in an email to the list. If you > search for Sparc in the mailing list, you should be able to find it. If > not, drop me a line and I'll see if I still have it. > I've got an Ultra 30 sitting here doing nothing. I'll see what I can come up with for Linux/Sparc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911
> FYI there is a way to do 911 its called E-911 enhanced 911 > the user has to set it up with the local emergency services > to it and you setup your pbx to xmit the data. There's PS/ALI (Private Switch Automatic Location Information) that's quickly becoming state mandated for all PBX systems. The problem with it is if your customers are spread out across multiple PSAPs or RBOCs. Then you've got to interface with the PSAP or RBOC in every area where you've got a customer. Then you've also got to assign a DID to every customer that can be transmitted back to the PSAP over the PRI or CAMA trunks (which are necessary to use E-911). Its fine if you're limited to one or two PSAPs in your service area, but a company like Vonage or NuFone has an almost unlimited number of PSAPs in their coverage area. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] far end disconnect supervision
> If some channel banks don't support this, how on earth do they know when > the telco side of the call has hung up ? They don't. They rely on either a timeout or the called party hanging up to disconnect the call. ___ 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] T1 Sync clarification
> Does anyone else have 2 t1's plugged into their T400 ? If > so, how are they synced ? This was just happening at night, > but I lost the second span a dozen times already today, all > within less than an hour earlier this afternoon. > If you've got 2 spans from the same provider, you should just be able to set them as sync type 1 and the other as sync type 2 (primary & secondary). If you've got spans from different providers...you're in for an adventure. You'll be able to do one of the following (which one is telco and luck dependant): 1) Set them up as primary and secondary sync and you might get lucky enough that both providers are synced close enough so you don't have to worry. 2) Set one up as primary sync, let the other ones derive clocking internally from the card, and deal with a bunch of frame slips. Are there any 1 or 2 bit buffers in the T1/E1 cards to allow it to resolve differences in clocking? ___ 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] How to Order Disconnect Supervision from SBC using Adit 600?
> Can anyone help me with the term that SBC uses to refer to disconnect > supervision? I have an Adit 600 channel bank which has helped improve the > disconnect detection time down to about 8 seconds. This is still causing > some > issues in particular with call progress enabled in * we are having a few > disconnects while calls are in session (about 2 reported in first 5 days > of use). > > I have talked both to a local phone contractor and SBC directly and no one > seems to know what I am talking about. The phone contractor knew about the > issue > with other phone systems in the area but didn't know there was a way to > fix it > and SBC reps seem to never have heard of disconnect sup or calling party > disconnect. I've never seen a line from SBC that DIDN'T come with disconnect supervision (some SBC line monkeys I know call it "battery drop disconnect"). > The * Handbook refers to loop start with call sup as kewlstart are > there other names for this protocol? One of the local contractors thought > that > SBC automatically drops line voltage on remote hangup, in which case I > need to > know what signalling to program into the ADIT 600's fxo channels. I also > have > the option of going to groundstart signalling if this would fix the > problem, but > it would cause some line downtime so it is not my preferred method. Kewlstart is also an alias for battery drop disconnect. > The Adit 600 manual lists the following options for mapping FXO ports to > the T1 DSO. > > DPT = Dial Pulse Termination > EMDW = E&M Delayed Wink start > EMI = E&M Immediate start > EMICPD = E&M Immediate Start with Calling Party Disconnect > EMW = E&M Wink start > GS = Ground Start > GSRB = Ground Start with Reverse Battery > LS = Loop Start > LSCPD = Loop Start Calling Party Disconnect > LSRB = Loop Start with Reverse Battery > VoIP = Voice over IP (CMG only) > > I believe I currently have the lines set to LSCPD which improved the > hangup > situation, but hasn't completely fixed it. That should be right. If you're really interested in looking, take a cheap voltmeter and put it across the line. If everyone is on hook, you'll see 48V. If someone goes off hook, you'll see it drop to about 6V. If you see a quick drop to 0V when the far end hangs up, you've got battery drop disconnect. > I don't know if this has any relevance but I am also originating the clock > source from the * side with Wildcard T1 card. That's really the only way it'll work. The channel bank can't generate clocking. ___ 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] How to Order Disconnect Supervision from SBC using Adit 600?
> I have a little more info on this. Following the suggestion of another > post on > this topic I tracked down an analog phone with lighted buttons powered by > the > phone connection. I directly connected the phone to one of my inbound > lines and > called it with my cell phone. Picked up the analog phone, verified call > completion and then hung up my cell. I watched and waited for the lights > to go > out. Sure enough they did, but it took 8 seconds from the time of the > hangup. > After the flash more phone started emitting a dialtone sound. Is this > correct? I > was under the impression the voltage drop would happen almost immediately. Do you have another analog line that's on the same Central office as the line in question? The delay could be lag time betwee the time you hang up your cell phone, the cell provider MTSO processes the hang up, passes it on to their termination provider, who then passes it on to your termination provider, who then passes it on to you. ___ 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] re: hardware requirement -asterisk
> # ifconfig xl0 > > xl0: flags=8843 mtu 1500 > > address: 00:01:02:78:11:e8 > media: Ethernet autoselect (10baseT) > status: active > inet 203.219.167.126 netmask 0xfffc broadcast 203.219.167.127 > inet6 fe80::201:2ff:fe78:11e8%xl0 prefixlen 64 scopeid 0x2 > > But ifconfig seems to suggest that it is running in simplex mode. If the DSL modem is running in full duplex mode and the card in your machine has auto-negotiated to half-duplex, things go to hell quickly. Auto-negotiation sucks anyway. It works about as well as Plug & Play. ifconfig xl0 media 10BaseT mediaopt full-duplex Or hell, try ifconfig xl0 media 100BaseTX mediaopt full-duplex Run "netstat -I xl0 -i 1" and watch for collisions. ___ 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] Grandstream transfer solution + DTMF translation possible?
> > The solution to the problems with the Grandstream 1.0.4.39 firmware is > to use inband (in-audio) DTMF. Neither the RFC2833 nor INFO seem to > work. Don't the Grandstreams send a DTMF 'F' INFO message on a hookflash? Shouldn't be that hard to change chan_sip to register an 'F' as an AST_FLASH control message. I'd do it, but I couldn't test it...since it seems that Grandstream hasn't found it in the goodness of their heart to add hookflash support to the HandyTone ATA. ___ 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] Example of TDM20B
> ; FXS Port 1 > context=local > signalling=fxs_ls > usecallerid=yes > echocancel=yes > echocancelwhenbridged=yes > ; > ;FXS Port 2 > context=local > signalling=fxs_ls > usecallerid=yes > echocancel=yes > echocancelwhenbridged=yes Change the signalling here to fxo_ls. Its gotta match what's in zaptel.com ___ 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] Incoming Voice/Fax Discrimination?
> I'm evaluating * to replace the crap set of peered "smart" phones we > have now in our small office, but I haven't been able to find out about > this anywhere yet: I need to know if * can discriminate _incoming_ FAX > calls on a voice line and route them to a specific extension? Yes, it can. ___ 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] Adtran 750 DID question.
> Hello All, > > I've mostly solved my DID problem from a few days ago. Apparenly the > lines weren't configured properly. Now heres the next question. 12 E&M > wink lines from telco. I have them all plugging into an Adtran 750 with > FXS cards. The Adtran ports are configured DPO. How do I signal this > from Zaptel. I have them setup E&M in zaptel.conf and EM_W in > zapata.conf. They work, however, no DNIS info is being passed. Do I need > to signal these something different like loopstart or kewlstart, so the > DNIS info gets passed? I watch the Tx/Rx bits from zttool, and everything > looks okay coming from the Adtran. It looks like asterisk isn't winking > properly. I had a similar problem. I ended up setting the trunks to either just plain em or featd (I don't remember). I chased through the chan_zap source code and decided (maybe incorrectly) that asterisk doesn't look for DNIS digits in E&M Wink mode. ___ 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] Auto dial in Off Hook situation.
> Thanks John, > > > I think it is not that simple. I am not using a phone but a Cisco ATA. > > The scenario: - > > User--(Multitech VOIP MVP200 (FXS))--Internet--(Multitect VOIP MVP100 > (FXO))--Cisco ATA--Asterisk--Any extension Any reason you can't use the H.323 load for the MVP200? I've not tried it in a year or so, but it mostly worked last time I tried it. ___ 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] Internal Lines Dialing Out
>> > exten => _.,1,Dial(Zap/1/$EXTEN)
exten => _.,1,Dial(Zap/1/${EXTEN})
Gotta put the name of the variable in brackets for it to work.
___
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Re: [Asterisk-Users] The Smallest Asterisk Server Ever?
> Now, here's the real question: can you install it on a toaster? It builds and runs on NetBSD, minus the hardware part (for the moment)...so yeah. Asterisk on NetBSD/Vax. Hrm. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The Smallest Asterisk Server Ever?
> On Tue, 3 Feb 2004, Chris Albertson wrote: > >> "Smallest" Asterisk server? No. That old Gateway box must >> be about 2 cubic feet. 1.5 ft^3 at a minimum. I've got one >> that is about 0.2 ft^3 a factor of maybe 10 smaller. > > Hehehe.. As far as "Form Factor" goes, I'm sure there are smaller boxes > out there. How about "Most resource challenged Asterisk server ever? :) > How about one of the 1Ghz Soekris boards with a 802.11 board in it? ___ 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] Newbie Question. Is asterisk right for my scenario?
> Hi, > > Please excuse me if my question seems too simplistic. I have been reading > the mailing list for some time and I am still a bit confused. Here is the > scenario that I would need to achieve and am wondering if asterisk is the > correct software to use. > > (h323) (h323/SIP) (h323) > pstn---cisco--Asterisk??-cisco---pstn > | > | > | -sip phone > > > > I have an existing h323 structure doing h323 pstn termination and would > like > add sip to part of the structure, also at the same time would like asterik > to act as a softswitch to store dial plans and make routing decisions. > Asterisk at the same time will do h323/SIP translation. > > My question, can Asterisk do all these? Or am I totally off? Quite simply, yes. Asterisk is a softswitch more than anything else. And it can take an incoming call on any of its available protocols (PSTN, IAX, SIP, H323, plus many more) and route it back out any of them. ___ 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] T100P & Phones Configuration
> Below you will find, what I believe to be a typical setup with a T100P > card. My question is - > > 1. Is this correct? Possibly. Depends on if you use a channel bank that can do add/drop and you're not using a PRI. You'll take your incoming T1 and go into 1 T100P and use another T100P to feed out to your channel bank...or you can get a T400P and just have one card in the system. > > 2. What kind of phones would be needed here... (Would you have to use > Digital phones) And if so what would you recommend. > You can use anything from a $9 WalMart phone to a $300 ADSI analog phone. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T100P & Phones Configuration
Exactly. > So... > > I would need as you noted two T100P cards or a T400P. The T1 goes into > the * Server and the second port of a T400P goes back to the asterisk > server. Then the extensions get broken out from the Channel bank? > > Geoff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T100P & Phones Configuration
Either way will work. Getting the T400 four port card gives you room to grow, but getting 2 T100P single port cards saves you about $500. > Is this the only way to handle extensions... This turns a 4 port T1 card > into a 2 port card... Is this the suggested method? > > Geoff > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T100P & Phones Configuration
> Ok.. Let me pose a question regarding this configuration. > > Lets say you have the ISP bring in a full T1 and they split it half > voice half data. They would usually do this in a channel bank on > site... So in this scenrio... You have the Channel Bank from the ISP > where they split the channels. Then from that it goes into the asterisk > for 12 channels of voice. Then from asterisk to another Channel Bank to > break out the fxs extensions Its not so much a "channel bank" that they use, but more of a CSU/DSU with drop & insert capability. The incoming T1 will go into the D&I unit, the data channels will get routed to whatever interface they'll be using (Usually V.35 serial) and the rest of the channels will go back out on a DSX port that gets connected to one port on the T400P (or one of the T100Ps). The second T1 port (or second T100P) connection runs to your FXS channel bank. > > It appears that the T1 Digium cards can split voice and data, but I > would not want data traffic going through the * server... Yes, they can do this. You can turn your * server into both a PBX and router. > > Geoff > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PrePaid Application!!!!!
UnixODBC. No need to rewrite everything for a simple DB change. > In what language is it written in? It would be interesting to at least > look at it and maybe convert it to use MySQL instead. > > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz > Jozwiak > Sent: Monday, October 13, 2003 3:49 PM > To: ASTERISK USERS > Subject: [Asterisk-Users] PrePaid Application! > > > > Hello, > > > > Here in our office we are testing Asterisk. > > My collage Igor created to Asterisk PrePaid application with Postgresql. > > It is not in Perl. > > We would like to release it to the group as soon as it will work > ok. > > It will have authentication, different rates for users, different rates > for destinations and so on. > > Is there anybody who would like to improve it ? > > > > -- Bart > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] "Gates steps up telecom campaign"
> On Mon, 13 Oct 2003 14:01:00 -0400, Andrew Kohlsmith wrote: > >>> I really wouldn't like to run a telecom system on Windoze in the first >>> place.. Last place I worked, we had to reboot our phone system every friday night...it used NT. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More beginner questions...
> Questions ... > > OK - So, I've got Asterisk up, a Cisco 7960 talking to it, some mailboxes, > and extensions. All exciting. > > Two questions: > > I'm in a natted environment and need to utilize a SIP provider to make > calls > in the US. Currently I have Vonage in my natted network and it works > fine, > however I understand there is no real way to make asterisk talk to Vonage > because they have a closed system. So, the question is, what SIP provider > do I go with? Does it have to be SIP? Several providers will do this over IAX...nufone.net, voicepulse.com > Second, I have 1 phone line coming into the house and I would like it to > be > routed through Asterisk. Is my best choice for this, a modem? Or, is > there > other hardware I should consider? X100P. Consider nothing else. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] We are thinking of Asterisk
> > Hi all, > We are thinking of changing our Nortel Meridian PBX to Asterisk. Before > we > jump into this we would like to know if we can support some important for > us > functionalities on Asterisk. We would like to know if we can > > 1. Have menu based voice mail with Asterisk? (like press 1 for sales, > press > 2 for tech support ...) Yes > 2. If we can login, logout make ourselves unavailable with Asterisk ACD? > If Asterisk will route alls FIFO bases to the loggedin and available > personal. Yes > > 3. Can we have regular PBX functionalities > Call Forwarding > Call Transfer > Conference Call > Music on Hold Yes > 4. Can we point DID directly to the extension? No need to dial a main > number and than an extension. Yes > > 5. Is it possible to monitor a call in Asterisk for training or coaching > purposes. If a new person is doing a tech support can we monitor (listen > to > conversation)? If yes, would it be possible to record conversation - like > a > voicemail? Monitor, yes. Recording...maybe > > > 6. Can we get our voicemail in our e-mail mailbox? (Can we delete, make > new, forward voicemail from e-mail client? > Yes > > 7 Can we send and receive faxes from PC? That's a work in progress. Sort of works, sort of doesn't. > > > > Thank you in advance for taking time and answering our questions. > > Aram ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI problems
I've got a couple of PRIs coming in from a SUMA 4 switch with some 800 numbers routed through it. When the calls come in, I get the following message on the console and the call never makes it through: (800 number is fake) Extension '8005551212' in context 'nonauthenticated' from '232102749585' does not exist. Rejecting the call on span 4, channel 1. I do have the following line in extensions.conf in [nonauthenticated] exten => 8005551212,1,AGI,ivr-main.pl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI problems
*CLI> show dialplan nonauthenticated [ Context 'nonauthenticated' created by 'pbx_config' ] '8005095639' => 1. AGI(ivr-main.pl) [pbx_config] '8005095640' => 1. AGI(ivr-main.pl) [pbx_config] '8005095641' => 1. AGI(ivr-main.pl) [pbx_config] > check 'show dialplan nonauthenticated' > > regards > Martin > > On Fri, 21 Nov 2003, James Sharp wrote: > >> I've got a couple of PRIs coming in from a SUMA 4 switch with some 800 >> numbers routed through it. >> >> When the calls come in, I get the following message on the console and >> the >> call never makes it through: >> >> (800 number is fake) >> >> Extension '8005551212' in context 'nonauthenticated' from '232102749585' >> does not exist. Rejecting the call on span 4, channel 1. >> >> I do have the following line in extensions.conf in [nonauthenticated] >> >> exten => 8005551212,1,AGI,ivr-main.pl >> >> ___ >> 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] PRI problems
It seems that there's a non-printable character at the beginning of the DNIS stream I'm getting from the SUMA 4 switch. Once I chopped that off, everything works right. > Hi James, > > Try to do > exten => _8005095639,1,Agi(ivr-main.pl) > > > Quoting James Sharp <[EMAIL PROTECTED]>: >> *CLI> show dialplan nonauthenticated >> [ Context 'nonauthenticated' created by 'pbx_config' ] >> '8005095639' => 1. AGI(ivr-main.pl) > ___ > 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] was FXO cards
>> Case 1 and 2 are ties in my eyes, except the channel bank would >> provably be cheaper to upgrade to 8 lines. I am just afraid of the >> channel bank. I just don't know anything about them. If I buy the >> wrong crap, it gets really expensive fast, plus adds another layer of >> complexity. You could also talk with your local phone company and other CLECs to find out pricing on fractional voice T1s/partial PRI. Depending on the locality, the breakeven point is usually 6-8 lines. You might even be able to get a deal on a hybrid data/voice circuit. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a problem with MeetMe
> The only problem I can think that you would have with the > ztdummy would be that to used a kernel source other > then the one your running when you build it... Or its not playing well with your USB hardware, which is what ztdummy uses to generate the 1Khz interrupts that zaptel needs. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call parking - Still haven't solved
> parkext => #700 ; What ext. to dial to park Try removing the # ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Line is stuck off hook...
Make sure you're using fxs_ks signalling for the FXO channels and also make sure that your incoming lines support disconnect supervision. Otherwise, * has no idea when the calling party hung up. > Hi Steven, > > I have analog lines connected to the fxo lines of the Zhone channel > bank. All of your suggestions sound good. How do you set up the config > file so it would play the greeting twice and then hangup the line? > > That would fix the problem for the most part, but why isn't * releasing > the line when the caller hangs up. > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX over IAX
The way I've seen it done is that the incoming fax signal is digitized and compressed, then sent over the IP channel. It is done in real time. You end up taking up 7k-14kbps instead of the 32/64kbps you'd use to pass high enough audio quality to not irritate the modems. Unfortunately, this takes the same DSP work that is necessary to provide fax transmission/reception...and there have been problems with making that work. > > I think the real solution is some piggy backed protocol that can be told > this is fax information at one end, digitize the fax as if it were a > faxmodem, stream it to the other end using a non-realtime protocol, and > then initiate a fax call at the other end and restream out the data, all > while possibly holding open the original call to indicate reception > confirmation at the end. > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CE certification for Europe
> On Thu, 2003-04-03 at 10:55, Mark Spencer wrote: >> > 1. it was FCC, CA and CE certified (FCC and CA states no card is reg >> with them as of last week) For the T100P and T400P: http://www.part68.org/tte_details.cfm?cicHistid=36439 http://www.part68.org/tte_details.cfm?cicHistid=36442 They were listed under "Linux Support Services", Digium's old name. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] small office
> On Fri, 2003-06-06 at 16:09, Dante Alzamora wrote: >> What is the best cost effective solution for a small office: >> I need 3 FXS & 2 FXO. >> >> Can I hookup a TDM400P and 2 X100P on the same computer? >> >> Also, I saw some IP phones for $25.99 >> http://www.wosmile.com/cgi-bin/view_store_item.cgi?pid=4424&sid=5&category=1 >> Can I use them with asterisk? will they be able to do the same as the >> TDM400P? >> I read that to run the conference app Meetme you needed a Zaptel >> driver. > > My BS detector is running high on that phone, It mentions dialing IP, > but there is no ethernet port listed or shown on the device. Also it > seems odd that you have to use 2 AA batteries for the calculator when if > it ran off of the phone line, there should be enough power there, and if > it was ethernet based, you'd be supplying plenty of power. Nope, not an ethernet phone at all. Looks like someone was trying to scam them off as IP Phones. They're just flashy chrome plated analog phones that should, in theory, work with the TDM400P. To answer your first question, yes you can have a TDM400P and 2 X100Ps in your computer. You just need to make sure they each get their own interrupts on the PCI bus or you'll get weird juju happening when you try to run the MeetMe application. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any plans for a .....
> On Sun, 1 Jun 2003, Gene Kochanowsky wrote: >> I saw that Digium now has a 4 port FXS card. Any plans to add a 4 port >> FXO card? > > I'd like to see a USB FXO... > With as much weird and wacky stuff that people have been seeing with the USB FXS (like totally not working, working only after a remove/insert cycle, working fine for 2 weeks & then requiring a remove/reinsert), I'd not even bother with the development time. Just my opinion...personally, I'm waiting for FXO and would also love to see an 4W E&M module as well. So many PBXs that use E&M in their tie trunks...so many applications for *. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Packet8 VOIP service now 1/2 the price
> > Too bad noone makes a cheap ethernet FXO. Why is it always FXS... :-/ > I've been wondering how hard it would be to make a cheap FXO device out of a Dallas Semiconductor TINI board and a TI DSP56000 chip. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
