Re: [Asterisk-Users] Don't know what to do if second ROSE component is of type 0x6
Tom Rymes ha scritto: Has anyone figured out what this message means: Don't know what to do if second ROSE component is of type 0x6 IIRC, it has to do with rerouted/forwarded calls. I came across that portion of source code when I dealt with call forwarding/deflection. "ROSE" stands for "Remote Operation Service Element"; some related information is encoded in information elements as a kind of "remote procedure" invocation/response, and the support for these things is in libpri is, as far as I know, not complete :) Bye, -- Emanuele ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN card
And now in English? ;-) I'm extremely sorry, I wanted to reply directly in order to cut down on FAQ traffic, but after I realized my mistake I did not hit "Cancel" quickly enough. Shame on me :) It boiled down to "try chan-capi with AVM, try bristuff with the QuadBRI and, at least in the latter case, find excellent courtesy start-up example configuration files included" :) (And, yes, I know, with this I'm polluting the ML and its archives with not one, but two useless postings!) -- Emanuele ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN card
Andrea Bencini ha scritto: I would like to know if somebody has already used a C2-ISDN or a 4BRiJUN card to connect asterisk to PSTN network. My problem is that i can't configure my asterisk. Please, help me with some solution! Non so se riceverai qualche risposta in lista, ma sicuramente lo hanno fatto in molti, sia con l'una (se è una AVM) che con l'altra! Qualche dritta per iniziare: con la AVM ti servirà chan-capi. Io ti consiglio il fork -cm mantenuto da Armin Schindler, disponibile su Sourceforge. Con la scheda quadBRI invece devi usare le patch bristuff, con i cui esempi di configurazione dovresti poter partire immediatamente. Saluti, -- Emanuele ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] setting up asterisk as an sms central?
Roy Sigurd Karlsbakk ha scritto: hi is it possible to use asterisk as an sms central to send SMSes directly to clients on PSTN instead of just communicating with a central? the telco to which we're currently connected doesn't have a central Yes, as far as you can spoof the Caller ID ;) The trick is that PSTN clients decide whether an incoming call is a SMS or not *before* answering, by looking at the Caller ID, and they are usually pre-programmed with the SMSC's phone number. (At least, that's valid for the SMS-capable analog cordless phones I've seen till now.) So, that's going to be a problem, unless your telco is willing to help you at least in that respect, and let you send a valid SMSC's phone number as caller ID. (Of course I haven't tried this across the public network, but I'd be ready to bet one or even two beers that it works!) Bye, -- Emanuele ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to call some numbers with I4L
Massimo Frisoni ha scritto: I have an EICON DIVA PCI 2.02 with I4L. I'm unable to call some numbers, in general numbers with automatic responders that do not rings. It's seems asterisk does not understand that the other party has answered, so after a timeout it reports 'busy', but in real the other end has answered. Any other call to a "normal" number works fine. You should probably try chan_capi and enable inband call progress reporting. (IIRC, that's done by putting "b" in the dialstring before the dialed number!) I wouldn't know how the same thing is done through I4L, maybe somebody else can shed light on that... Bye, -- Emanuele ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr server
Altus Snyman ha scritto: Good day all Is it possable to set asterisk up as a cdr server for other voip units We got a quintum dx here and its got a option to log to a cdr server on port 9002 That's not a job for Asterisk! The Tenor can connect to a server and send CSV records over TCP, so you may want to write something (a short script may do) that interfaces to your existing billing system, inserts each record into a database, or — simple option, but not very useful — copies the CSV lines somewhere for later use. Bye, -- Emanuele ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax phone and asterisk server on different LANs
gincantalupo ha scritto: I have a *IAX* phone connected to a LAN and I want to connect to it to make calls using an Asterisk server located inside another LAN behind a router (* hasn't a public IP) I bought a IAX phone because SIP had problems with nat and so on. ::)) How should I configure the IAX phone and Asterisk in order to make calls? There should be no need to change the Asterisk configuration, IP-wise. Just configure the phone so that it knows the public IP address of the router on the remote LAN, and set up port forwarding on that router so that the IAX port is forwarded to the Asterisk box. > Is it possible without touching the router configuration? Not without sorcery! :) Bye, -- Emanuele ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on AMD64
Massimo De Nadal ha scritto: I think you can't run asterisk on a 64 bit linux version. Certainly it works well on a 64 bit amd processor, but in x86 mode... Am i right guys ?? It does work all right here as a native 64-bit app, on a 64-bit kernel with zaptel and BRIstuff! Greetings, -- Emanuele ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_sms: using * as an smsc
Tobias Wolf ha scritto: Let us assume that i have a couple of phones which should be able to receive SMS directly from my * box ( and not from an SMSC from BT or Deutsche Telekom ), So all these phones have the phone number of the * as Service Center configured. I recognized that the numbers of other SMSCs differs for outgoing and incoming SMS. I tried that successfully with my own SMS rig a couple of years ago. As far as I could tell from experimenting and from the ETSI docs, the phone knows it shouldn't ring, but it should answer and talk FSK to the SMSC, by looking at the caller ID; so, yes, you should set the correct caller ID in * to talk to your phone. Regards, -- Emanuele ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is it possible to use CHAN_CAPI with ZAPHFC enabled card ?
Armin Schindler ha scritto: I don't think so, unless someone has written a CAPI layer for HFC-S PCI A cards! Isn't mISDN providing this? Right. It didn't work for me, but a lot of time has passed since I tried it! Thanks, -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is it possible to use CHAN_CAPI with ZAPHFC enabled card ?
[EMAIL PROTECTED] ha scritto: In the end I succesfully compile zaphfc, but I am not able to use the card (a lot of problem running zapcfg, a loto of problem starting asterisk saying about wrong anything (from signalling to any other parameter specified in zapata.conf) You may want to post both the configuration files AND the error messages here... is it possibile to do not use zaphfc and configure in some way a CHAN_CAPI channel pointing to Billion card ?? I don't think so, unless someone has written a CAPI layer for HFC-S PCI A cards! Bye, -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need to start from somewhere
Darko Sundek wrote: > SORRY FOR MY LONG PRELUDE ( we respect kbps) Molim, pozdravi mi Podgoricu i celu Crnu Goru :) >1. What we need to know about our LOCAL PSTN telco (digital) lines to > be shure in our hardware choise (voltage, current etc.)? *IF* they are EuroISDN, then they are standardized, at least at the physical layer, and you shouldn't worry about it. I'm quite confident that they are, since the national operator must have bought standard equipment for the central offices. >2. We need 4 PSTN line system which can recive and make calls (on all > four lines) from and to the local VoIP network, hardware > sugestions PLEASE? You could use one Digium TDM04B, which is a much better idea than four separate cards. Otherwise, you could look for external adapters, but I'm not able to tell you which are both cheap and reliable! Bye, -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone have success with BRI in Italy?
Giorgio Incantalupo ha scritto: Hi, the question is: can digium and quadBRI co-exists easily on the same server? We are still having a lot of troubles since it is hard to find infos on how to configure them. I think they can. At least, I have HFC-S PCI A cards and a TDM analogue card, rev. H, working together without any noticeable problem (no data/fax traffic). With the QuadBRI it should be even easier. Bye, -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATXFER discussion, what's your opinion ?
[EMAIL PROTECTED] wrote: > By the way: anyone got experience in attended trasfer with snom ? :) Works OK here, using a lightly patched CVS from a couple of months ago and the instructions that they provided (HOLD, dial extension, speak to said extension, then TRANSFER). Of course this isn't going to work nicely when you have more than one call on hold, so I'm looking forward to testing the patch that Frank Sautter wrote about on these lists some days ago! Bye, -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone have success with BRI in Italy?
Kevin Hanson wrote: > Can anyone recommend a BRI card that supports Asterisk and that will > work in Italy? Will the Digium TDM card work in Italy? I guess that everyone here will recommend the quadBRI (e.g. Junghanns'). Digium's TDM card does not support BRI! -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quadbri trouble
[EMAIL PROTECTED] ha scritto: Hi, Thank you again for the help is there a way to debug the isdn ? Yes: 1. cat /proc/zaptel/* will provide hints about Layer 1 (ACTIVATED and DEACTIVATED are self-explanatory, but the code that comes with them isn't: you will find its meaning on the HFC-S PCI A datasheet that can be downloaded off www.colognechip.com); 2. from the Asterisk console you can use "bri debug" and "bri intense debug" to watch the Layer 2 packets. Bye, -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quadbri trouble
[EMAIL PROTECTED] ha scritto: I know the configuration of telco is point-to-point and I think the card have to work in NT mode (I presume because I have not found the documentation about this and when attach to the ISDN the led become green). If you're connecting the card to a NT1/NT1+, then *that* is the NT (network termination); your card has to work in TE (terminal equipment) mode. The card must be in NT mode when * has to work as the "network" side (you were mentioning a Hipath PBX a couple of weeks ago...) signalling = bri_net bri_cpe pridialplan = local prilocaldialplan = local They'll probably have to be "dynamic" and "unknown". Bye, -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Enable verbose output for TxFax/RxFax
Stefano Arata wrote: > Hi, I'm using asterisk with a digium TDM400P: I can't send/recive faxes > with a Philips fax machine. > It seems that the fax machine doesn't recognize the carrier. I'm afraid that your problem is not spandsp, but the TDM400P. Take a look at this (long) thread: http://lists.digium.com/pipermail/asterisk-users/2005-May/105081.html You may also want to take a look at Steve's FAQ list: http://www.soft-switch.org/spandsp_faq/index.html -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] annoying static when calling from legacy PBX -> * ZAP interface
Bernie Ott wrote: > http://www.auerswald.de/int/products/c4410usb.htm ) which I connected > to my 2nd ZAP interface (s0 <-> Zap) via Crossoverr ISDN cable (which > I crimped myself, I guess that's not the source of my trouble). What hardware? What driver (zaphfc?) Have you got any messages in syslog? -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fw: Multiple Quad Bri card
[EMAIL PROTECTED] wrote: > > Hi, > I would like to connect my * with two quad bri card: one to my Hipath > pbx and other to the telco. > I successfully installed the cards to asterisk patched with bristuff, > now how I tell asterisk that I have 2 qud bri card. The driver will recognize them. Next, you edit zapata.conf and zaptel.conf in the same way you would set them for an OctoBRI (I believe that there are example config files in bristuff!) -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!
harry gaillac wrote: > I agree you. > > Does asterisk (Digium) project provide a good > documentation ? [...] If you think that all big IT corporations are virtuously advancing technology for the benefit of us all, with no exception, and that the OSS community is a bunch of worthless scums trying to make money off other people's work, then please go buy those enlightened firms' equipment and software, and avoid asking for help on the mailing lists of OSS projects like Asterisk, SIPFoundry, Vocal, GFax, or similar forums. You will surely find that those corporations provide wonderful, timely, infallible tech support that will guide you through each step, without asking you for any extra money, and you won't have to waste your own time on Google (yes, your time is precious, while the developers' is as worthless as themselves). And you won't even have to ask for unrelated help on the IETF SIPPING mailing list. ;) -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Loosing hair on connecting Panasonic PBX- *-Euroisdn Italy
Robert Rozman wrote: > I had this experience with original company No answer for 14 days... > > So I got a little precausious, how would SW-drivers support look like, > if someone even doesn't want to sell HW... Well, at least they wrote them :) Anyway, a bri [intense] debug is in order to help you on the dropped calls problem :) -- Emanuele (from Videm ;) ) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr and billing
Mahmoud Badran wrote: > thanks alot for help but problem is; consider this scenario an internal > sip phone calls the IVR which shouldnt be billed then he dial an > extension from the ivr that redirects him to outbound line that makes > the call have some time counting in the ivr and other time counting > during the outbound call so how can i bill him on the outbound only?? ForkCDR and ResetCDR will probably help you! -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Loosing hair on connecting Panasonic PBX- *- Euroisdn Italy
Robert Rozman wrote: > I wanted to do this (it's principle I always follow) , but we even > haven't received offer to pay for the stuff (we applied twice for offer > of two cards), so bought where we actually could buy something... A customer of mine has had the same problem with the Italian dealer: they behaved as though they didn't want to sell :( > What is your experience of authors of software ( I guess we all know > whom we talk about) Here I don't see exactly what you mean. I referred to Klaus-Peter Junghanns, anyway! -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Loosing hair on connecting Panasonic PBX- * - Euroisdn Italy
Robert Rozman wrote: > I'm pulling my hair down and getting bold :-) . I have Asterisk between > Panasonic KXTD816 and Euroisdn in Italy (beronet octobri and bristuff > Asterisk) (hint: spend the extra $$ and support who's written the software!) > I'm trying to do just plain transfer of call from pbx to ISDN through > Asterisk... I'm doing that without any problems via normal HFC-S PCI A cards with Samsung PBX's. > It seems like PBX hangsup, when call is progressing with no apparent > reason. > I'd kindly ask for any advice or some working example for this Would you mind checking if Layer 1 is UP (cat /proc/zaptel/*) and reporting "bri debug span ..." traces? > On isdn side I also have a problem. Asterisk quite often says that it > cannot create ZAP channel, although partticular span is reported up and > active. I've also tried to connect loop between NT and TE port and > call doesn't get through So it looks like it does not depend on the Panasonic gear! -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: New JAVA application server for Asterisk - OrderlyCalls
Matt King wrote: > The reason for this is that Orderly Software provides an advanced queue > management system called OrderlyQ, that lets callers hang up and call back > when they reach the front of the queue. OrderlyQ is patent-pending, > and we do NOT allow the use of OrderlyCalls to provide similar > functionality. I'm quite curious about how this could be patented, since it's already happened to me quite a few times to run into queuing systems that call me back when I reach the front of the queue. Wouldn't that qualify as prior art? (I'd have many more grounds to dislike the notion that it could be patentable, but the "prior art" one is one where my viewpoint and the law's might agree!) -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-biz] sipredirect question
Axel Schemberg wrote: > I use Asterisk on Debian via: ap-get install asterisk, which is Version > 1.07. The page you linked says: "new in Asterisk 1.2.x". I guess that this pretty much explains why it does not work in your case :) BTW, this looks like a -users question to me, so I've moved it there. -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Console ALSA Sound
Sean Kennedy wrote: > Heh, try asking about line appearances and the hint priority. People > clam right up. Or ask about receptionist phones that show all your line > statuses. > > You can practically hear the crickets. :) A quick search through the ML folder told me you asked about that on May 31. I've set up line status on Snom phones a few days ago, using HEAD from May 29 and bristuff, and I didn't have any luck until I realised that the support was broken in pbx.c: maybe that was your problem too. It has been alredy fixed in CVS :) I haven't tried with stable. -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] EuroISDN Italy - quadbri - zaptel.conf - whatsettings work ?
Robert Rozman wrote: > Is framing and coding (ami,ccs) right for Italy ? They are dummy settings with bristuff. The example config will surely do :) -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] rxfax not answering
Antonio Gallo wrote: > the call is originated by a FAX on PSTN and received via VoIP by > asterisk using a/u law codec It must be said that such a setup is not *supposed* to work in the first place, although it *may* work: http://www.soft-switch.org/foip.html (Steve, I'm saving you one post... ;) ) That said, you are supposed to hear something if you dial normally into that extension from a normal phone. If you do, but dialing from a fax does not achieve the same result, then I guess that your provider is handling fax calls differently! -- E. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: ISDN 4 BRI card for UK
Leandro Morgado wrote: > How did you manage to get 4 Fritz cards in the same box? Could you give > details on what kernel and kmodules you used? Which asterisk channel > are you using? chan_capi? chan_misdn? One way of doing it is: http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does Debian Bristuffed Asterisk work ignore Beronet cards ?
Robert Rozman wrote: > when loading qozap it says that no multibri card was found although > lspci shows it... There were quite some rumours about bristuff not > liking other than junghanns cards, but don't know if something happened http://www.beronet.com/download/card_installation_guide_en.pdf On page 37 you'll find that bristuff must be patched in order to recognize other cards. > Anyone recently used this card and Debian Asterisk and can confirm that > this is working ? Any advice or hint, what should be done to get it into > working state ? I've never tried, but in similar situations I've changed the PCI ids in the drivers. This will surely make any driver recognise the card; it doesn't mean that it will surely work, though :) -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc: empty HDLC frame or bad CRC received
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Marco Parmeggiani wrote: > hi, what do you think? this is a bit too much low level for me. You aren't sharing it -- that's fine. I would guess that the cable is not OK or, even more probably, the bus is not correctly terminated - but then again, if it works with the other drivers, it doesn't look like it. I hope somebody else may be more helpful than me :) - -- Emanuele -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.0 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFClfJqdh0IFKddsi4RAnyTAJ9GtpVDqqdCmI3/q9+zB0GO/H7GIQCgsL5y iEzSFeU1FqdgMY4VrmvKr44= =hwc/ -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc: empty HDLC frame or bad CRC received
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Marco Parmeggiani wrote: > Hi, i've downloaded/compiled/installed the bristuffed asterisk > Asterisk 1.0.7-BRIstuffed-0.2.0-RC8a > and i'm using it with an hfc card. It runs on a debian 3.1 sarge machine > with kernel 2.6.11. Asterisk works well if i configure the card using > isdn4linux. Are you sharing the IRQ? (check /proc/interrupts) - -- Emanuele -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.0 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFClNTBdh0IFKddsi4RAvQTAJ9C9rnaBZ9ekbECQM8l/+gVNIFVuwCbBQtb Mw3nVzlGh2fFuHz0DoG9xOM= =fYUh -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] two isdn cards
Stankiewicz Michael wrote: > thanks a lot, > i've googled around hunting for an answer to my biggest doubt: the > cross-cable. > i understand that it looks like an cat-5 cross-cable and how it has to > be done, but ... why 8 wires ? The plug is a standard RJ45 one, but only the 4 inner wires are used. Sometimes you need the outer wires to carry power, but I believe you shouldn't cross them. I strongly suggest that you do NOT use the T1 cable schemes, since they do not have much to do with the S0 BRI bus. The link I mentioned in the last message (isdn.jolly.de) carries the information you need, and I can assure you it works (without even the NT). If you connect a "live" NT (hooked to the central office) to Asterisk to access your ISDN line, then you are going to use a card in TE mode, without any special configuration, and a normal, off-the-shelf ISDN cable. Regards, -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] two isdn cards
Stankiewicz Michael wrote: > i followed this how-to: > http://www.voip-info.org/tiki-index.php?page=Asterisk%20zaphfc%20install26 > having in response no sign of life. If the module doesn't even get installed, or the kernel does not report any card as recognized, you could tweak the initialization routines to add PCI IDs for your own cards, and hope they work correctly. If the cards are recognized, there should be nothing to worry about: either they work with zaphfc or they don't, modulo interrupt troubles. > the software side is pretty straightforward but i have many doubts on > the hardware deployment: > 1- the idsn cable going from asterisk to the NT sould be a cross cable ? Yes. But not an Ethernet cross-cable, an ISDN cross-cable; there's a pointer on the wiki to a page on isdn.jolly.de explaning how the cable should be made, and suggestions about how to take advantage of a disused NT. I reckon that telephony folks call it an "ISDN TX/RX" cable. > 2- it should have 100 ohm resistors (if yes, where can i find the > schemes )? Yes, the bus should be terminated (so the resistors don't have to be on the cable itself). A full description of the bus is in the ETSI standard for ISDN layer 1 (www.etsi.org). -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tiny install with Solid State Storage
Il lun, 2004-03-01 alle 18:14, Andrew Kohlsmith ha scritto: > Has anyone looked at using busybox and uClibc with asterisk? Those two (and > agressively stripping everything) were the biggest things in making Linux > tiny. That and eliminating static binaries whereever possible. I've run a uClibc-linked Asterisk in a chroot environment for some months now, and I have to say it works. It isn't stress-tested, but it seems to work. The binary is 546 kB, and all 90+ modules take up 2364 kB together. -- Emanuele ___ 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] CTI/TAPI
Hello, > But to answer your question, I have a friend that does Checkpoint firewall > training/consultation and he gets upto $20,000 per week for running training > classes. Not in the US mind you but abroad, mostly in Europe. He says > American companies are too cheap. I wouldn't say anything about your friend, but my own opinion in a general case would be that whoever pays a single trainer $20,000 per week of training either does not understand the value of the money he's managing, or is getting back a proportional bribe. (I'm European myself.) Not that I wouldn't accept $20,000 per week - I simply wouldn't pay them (and, if I accepted them, I'd keep 100% of them.) ;) -- Emanuele ___ 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
Il lun, 2003-12-01 alle 15:36, Brancaleoni Matteo ha scritto: > Why not Sardinia, in Italy? > good food, nice people :) Since this thread has already grown way larger than it should, may I add Venice? :) -- Emanuele ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo cancellation
Il mar, 2003-11-25 alle 14:28, Peter Zeltins ha scritto: > Hi, > > I'm interested. I'm running chan_capi 0.3.0 with Fritz PCI ISDN card. Using > DIAX as softphone and dialing out to PSTN generally results in good sound > quality at softphone end (no echo), but PSTN end experiences quite a bit of > echo. I have enabled echosquelch in capi.conf, but it does not seem to help Then my idea is not your solution, nor is echosquelch, I guess. Yes, it would be possible to kludge chan_capi to alleviate your problem; no, I do not think it would be the right solution. If the far end hears echo, it's coming from the near end, that is from your softphone. You could try to use something else to confirm that (a hardphone?). I suggest that you play with the mixer settings on your own machine, because it's your sound card that's recording the far end's voice! Bye, -- Emanuele ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [Asterisk] GSM access
Il lun, 2003-11-24 alle 14:08, Jon Stockill ha scritto: > Maybe bluetooth would be the answer - have the pc register with the phone > under a headset profile, and you'd have your audio. Use AT commands on a > comms profile to dial? Reading the specifications, it seems that you need a CVSD codec to do that, so its implementation would probably be the first step. (Maybe some mobile phones, or most, or none support some more codecs over Bluetooth, I have never tried that.) Of course, I haven't tried anything myself; this is mere speculation. :) -- E. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo cancellation
Il lun, 2003-11-24 alle 23:19, Peter Zeltins ha scritto: > What is the status on echo cancellation in Asterisk/CAPI? I tried to straighten this out; I think it works, but I'm not sure. If it does work, I think it might find its way in future chan_capi releases; if you want to try it out at this early stage, contact me off-list! -- Emanuele ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tunnel iax via gnophone with ssh?
Il ven, 2003-11-21 alle 11:29, Fearghas McKay ha scritto: > At 15:38 -0500 20/11/03, Billy Huddleston wrote: > >Use CIPE, It's a UDP based VPN solution. > > Don't use CIPE, it has holes in it and is breakable. I've started testing OpenVPN and it doesn't seem to be that bad. Its main weakness, IMHO, is that it does not scale, but if you have a handful of sites to connect, it could do the job. -- Emanuele ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX/IAX2 encryption?
Hello, > The PGP documentation suggestes that users cary their key > in a floppy and never copy the key file to the hard disk. > So your "little black plastic key" is a floppy with the write > tab punched out. Maybe I've missed an important turn in this thread, but it seems to me that the discussion was about encrypting phone conversations when users are "on the road". Wouldn't using a floppy disk or a pen drive with your own private key on an untrusted machine defy the whole purpose of keeping it private? Probably it can be helpful anyway in most situations, and is surely better than no encryption at all, but it seems to me that a good solution to the problem implies some kind of smart encryption device (why not on USB, rather than a smart card!); that should be enough to foil also man-in-the-middle attacks, if at least one endpoint is already trusted. Bye, -- Emanuele ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] capi_chan error - CAPI not loaded.
Il mar, 2003-07-22 alle 20:44, Peer Oliver schmidt ha scritto: > System: Debian 3.0 / apt-getted most stuff except chan_capi and b1.t4 > Jul 22 19:47:03 NOTICE[16384]: File chan_capi.c, Line 2568 > (load_module): CAPI not installed! > Jul 22 19:47:03 WARNING[16384]: File loader.c, Line 299 > (ast_load_resource): chan_capi.so: load_module failed, returning -1 Two hints. 1 (the stupid one). Are you running * as root? 2 (the less stupid one). Have you got all the kernel modules in place? Here's an excerpt from my own lsmod: Module Size Used byTainted: P f2pci 540896 2 fcpci 540832 2 capi 18208 4 capifs 3980 1 [capi] kernelcapi 30304 4 [f2pci fcpci capi] capiutil 22816 0 [kernelcapi] (for those who wonder about f2pci: yes, not only am I cheating, but it SEEMS to work rather fine. Half a meg of RAM is not such a big price to pay, compared to that of an active card.) -- Emanuele ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] *--IAX--* problems. (chan_capi problem)
Il mar, 2003-07-22 alle 15:41, WipeOut . ha scritto: > Ok.. I have done some more digging and the problem seems to be caused by chan_capi > not detecting that the call has been answered.. I downgraded chan_capi from 0.2.3b > to 0.2.2 and the system is working fine.. I have tried to tackle that issue today. It seems that the problem of answers not being detected can be solved by uncommenting lines 1712 and 1718 in chan_capi.c (0.2.3b), but I haven't yet got a chance to test what REALLY happens then. :) It just seems to work, but further surprises might be just round the corner... -- Emanuele ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Remote Agents
Il lun, 2003-07-14 alle 20:48, Derek Barber ha scritto: > One of the key features we need is the Remote Agent, I am not sure how > this works and was wondering if someone could give me some information > on that. We would like to have calls routed through Asterisk to remote > agents at home and then have a screen-pop on their PCs that would give > details of the incoming call. We have an ISDN PRI connection through > which the calls will be routed. My 2 cents: someone claimed on a web (or I should say "wiki"?) page that he wants to write a XWT interface for Asterisk, so that it could run on any platform, and that he is working on a XMLRPC server interface for Asterisk. I'm afraid that he isn't anymore, though: the page was last changed on March 9. http://wiki.xwt.org/Wiki.jsp?page=JaysonVantuyl -- Emanuele ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] modules.conf again
Il mer, 2003-07-09 alle 09:31, carlos del mayor ha scritto: > THANKS VERY MUCH in advance, and here they are, my two > little questions... Well, here are my two little answers, I hope they are not too wildly incorrect :) > 1)As I have seen, to make Asterisk load chan_capi.so > and chan_modem.so you must have: load=>chan_capi.so > and load => chan_modem.so in your modules.conf. But I > had understood some time ago that setting autoload => > yes made Asterisk load every module that was necesary. > Then, why must I load these channels explicitely? Because that way they are loaded first. Some other modules use symbols that are exported by those modules, and if those other modules got loaded first, they wouldn't work. > 2)For what is used the section [global]? My sample cfg says: ; Module names listed in "global" section will have symbols globally ; exported to modules loaded after them. So it is needed to get the aforementioned result! Bye, -- E. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi error!!
Il mar, 2003-06-17 alle 18:59, WipeOut . ha scritto: > Have I left something out or done something wrong?? Yes, in modules.conf: [global] chan_capi.so=yes You need it in order to export its symbols to the applications! Bye, -- E. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi error!!
Il mar, 2003-06-17 alle 20:01, WipeOut . ha scritto: > I do have that in modules.conf > > Anything else? Yes, load=>chan_capi.so in the same file, but in the [modules] section. No other idea comes to my mind. :( -- Emanuele ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Speex
On Tue, Jun 17, 2003 at 02:27:38PM +0200, Tjardick van der Kraan wrote: > It seems * is not loading speex. When i did a make in the codecs sub dir, > the following error pops up when making speex: > > codec_speex.c:34:19: speex.h: No such file or directory > > is this file missing in the cvs as i just removed the whole * dir and did a > new checkout and still seem to get this error, or do i need to get/install > something before speex works ? Yes, you ought to install libspeex! It is probably installable by your distro (libspeex-dev in Debian, for example). Bye, -- Emanuele ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Installing the wcfxs driver
Il lun, 2003-06-16 alle 15:22, Robert Boardman ha scritto: > My Motherboard cannot disable the IRQ sharing, can I specify on with modprobe, > the IRQ to be used with a particular module? I do not know to what extent you can play with the kernel code in order to change how IRQ's are handled. Possibly none, but even if it is possible, I have no idea myself how to do it. Surely, though, that cannot be done with modprobe. Until now, my best solution to your problem has been moving the S400P board to a computer with a different motherboard. :( -- E. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Installing the wcfxs driver
On Mon, Jun 16, 2003 at 12:05:34PM +0100, Robert Boardman wrote: > for a couple of days or a couple of hours but then stop, I'm a complete linux > newbie, how can I force the wxfxs driver onto another IRQ in case it is this > causing the problem You usually can, you should check your motherboard's documentation. I have an Asus MB and I can effectively disable IRQ sharing for the board in the setup area reachable at boot. Bye, -- Emanueel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using Linux traffic shaping to prioritiseSIP/IAX traffic?
Il mar, 2003-06-10 alle 20:07, Stephen Davies ha scritto: > > When the "tos" option is set correctly (to "nodelay"), the default > > queueing in recent kernels already does that, because the pfifo_fast > > queue is used (if I recall correctly). > > But there is never any queue on my Linux box. It all storms out of > the ethernet interface and gets queued up in my cable modem which > doesn't know anything about tos settings. That is not entirely correct. There is an output queue, and pfifo_fast is the default (see the LARTC Howto, 9.2.1.1). But you are right when you say you need something to slow down the data;the simplest choice should be addingthe Token Bucket Filter (9.2.2.2). But if the wondershaper already does it all, then it's probably better to go with it... :) -- Emanuele ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using Linux traffic shaping to prioritiseSIP/IAX traffic?
Il mar, 2003-06-10 alle 17:19, Stephen Davies ha scritto: > Has anyone done anything with the Linux advanced routing stuff to give > SIP and IAX traffic priority? > > What I have in mind is a high-pri queue for voip traffic, all the rest > in another queue that gives way to the VOIP stuff. When the "tos" option is set correctly (to "nodelay"), the default queueing in recent kernels already does that, because the pfifo_fast queue is used (if I recall correctly). -- Emanuele ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call rejection with chan_capi
Hi, I remember that someone on IRC was curious about ISDN call rejection with chan_capi. I have to say that it works very well (thanks kapejod!). The reason why it didn't work at first was the "intelligent" NT1+ box - an ISDN NT with analog ports. You have to tell it explicitly that you want to disable the analog port corresponding to the MSN being dialed. If the analog port is disabled, and there is nobody "watching" that MSN on the S0 bus, the call will be rejected immediately. If the analog is disabled, and chan_capi picks up that MSN, then the calling party will get the normal ringback, and a congestion tone shortly after the call is rejected. If the analog port is enabled, then call rejection does not work. I guess that the "intelligent" NT wants every device to agree on rejection, or something like that. :) At least here (Italy), the result code for the calling party seems to be always 0x349F (Normal disconnect, unspecified), but I have not done a lot of experiments. Bye, -- Emanuele ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T1-PRI deployment questions...
Il gio, 2003-05-29 alle 23:10, Steven Critchfield ha scritto: > I still can't find any reference to AMI being lossy, and can't find any > comments that show where a AMI circuit would introduce 1's to maintain > 1's density. After reading a page describing test patterns and why they > use certain test patterns, it makes sense why AMI might not be usable > for a PRI though. I think that most uses of AMI imply "density enforcement", and do purposefully throw in some 1's here and there to break 0-sequences. Google found many matches, among which is http://www.laruscorp.com/chapt02.htm . Bye, -- E. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
