RE: [Asterisk-Users] zaphfc: empty HDLC frame or bad CRC received

Hi, Don't know if you've sorted this or not, but zaphfc: empty HDLC frame or bad CRC received (framelen = 5, stat = 0xff, card = 0). I've only ever seen when the signalling is wrong. For example if the line is in PTMP mode when it should be in PTP or vice-versa. this is the

[Asterisk-Users] Looking for wall mountable cases in the UK.

Hi all, I am in the UK and am struggling to find a decent ATX case which doesn't look like a PC case and can be wall mounted. For single BRI installations, I use the Pack-Box from http://www.icp-epia.co.uk/ which, along with an EPIA board works a treat. For multiple BRI or PRI installations, I

RE: [Asterisk-Users] PRI setup

Mike M: What I haven't found is a guide that gives the following steps: If you're in the UK, working with BT ISDN30, I've done a few installs, so drop me an e-mail and I'll let you know what I know. Nick. ___ Asterisk-Users mailing list

[Asterisk-Users] Goto invalid extension doesn't go to 'I' when in a macro.

Hi, A bit of a problem here which I'd appreciate some thoughts on. (please excuse the stray capital letters - Outlook has a habit of capitalising where I don't want it to!) For various reasons, I need to be able to do the following: --8-- [default] Exten =

RE: [Asterisk-Users] Intercept and redirect outgoing calls ?

I'd like to do simple LCR - when user dials number, would like to check against database and if that number is available over VOIP, simply substitute dialed number with SIP address for instance and call over VOIP. www.e164.org Nick. ___

[Asterisk-Users] Useful information - UK exchanges and SystemX/SystemY and how ISDN works!

Hi all, I thought this may be useful information for the Brits reading this list: I installed a * box (one ZapHFC BRI) for a customer last week and noticed the following warning every fifty seconds: == Primary D-Channel on span 1 down Dec 6 17:31:19 WARNING[213005]: chan_zap.c:1989

RE: [Asterisk-Users] Record() and problems converting with sox.

Kevin P. Fleming This means exactly what it says. You have asked for the output sample rate to be 8KHz, so apparently the input sample rate must _already_ be 8KHz, which means there is no resampling required. Indeed. This much is obvious. Since you said that the input file was

RE: [Asterisk-Users] Record() and problems converting with sox.

Kevin P. Fleming: In other words, to take a WAV file _created by Asterisk_ and turn it into GSM, this is all that is needed: $ sox file.wav file.gsm Which is where I'd got to. However, it doesn't appear to work - all I end up with is a very loud hiss So that's when I looked at the

[Asterisk-Users] Record() and problems converting with sox.

Hi, I'm trying to convert a high(er) quality wav/ulaw/alaw file (captured with Record()) to a gsm file and can't get the bugger to work. The example on the page http://www.voip-info.org/wiki-Asterisk+sound+files says that: $ sox inputfile.wav -r 8000 -c 1 outputfile.gsm resample -ql Should

[Asterisk-Users] Variable substitution - How can I do Dial(${DIALSTRING}) where ${DIALSTRING} is 'SIP/201, 15, tT'?

I've been banging my head against a brick wall for the last hour and I'm sure this is one of those easy to solve things - just that I can't see the wood for the trees. I'm trying to do: --- [some-context] Exten = s,1,Macro(dodial,'SIP/201,15,tT',123456,MOHClass) [macro-dodial] Exten =

RE: [Asterisk-Users] Variable substitution - How can I doDial(${DIALSTRING}) where ${DIALSTRING} is 'SIP/201, 15, tT'?

OK, sorry for the mistakes in new stack Nov 29 19:43:09 WARNING[802835]: pbx.c:1280 pbx_extension_helper: No application 'Dial{${ARG1})' for extension (macro-dodial, s, 5) Should have read: -- Executing Dial(SIP/201-aca6, SIP/201,15,tT) in new stack Nov 29 21:14:05

RE: [Asterisk-Users] zaphfc sound problems

Thomas Jagoditsch: my general impression was that too, but most users seem to work with the setup tim recommends. Having wasted a load of money on Fritz! Cards from eBay, I too had the problem with stutter and terrible audio with chan_capi. I moved over to ZapHFC cards and haven't had a

RE: [Asterisk-Users] 4 port ISDN BRI pci card

David Uzzell I want to be able to use 2 bonded ISDN BRI's and I am not sure what hardware will run with asterisk? Anyone got any ideas? I have a couple of customers with two HFC cards working on system access (PTP) mode with no problems whatsoever. The cards have the major advantage of

RE: [Asterisk-Users] Dial without bridge

Adam Greenbaum: Is it possible to dial a number (from an extension dial plan) and not bridge the call immediately? Yes. I want to announce something to the callee before the caller is connected. The A(x) parameter of the dial command allows this - see

[Asterisk-Users] Continuing a call to callee after caller has hung up.

Hi, I need to be able to continue a call with a callee after the caller has hung up. Is this possible using the 'Dial' command? The situation: We have inbound calls which we'd like to gather some stats on - audio quality of call, type of call (sales, support) etc. The best time to do this is

RE: [Asterisk-Users] Pause during dial

Henry Devito: exten = 3,8,Dial(sip/${destination}D{$pin}) ^^ Awoogah. Awoogah. Nick. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To

RE: [Asterisk-Users] failed to go to next dial command

MvB: Is this possible in Asterisk Yes. and what should be the approach? Read the Wiki ;-) http://www.voip-info.org/wiki-Asterisk+cmd+dial Look at the 'g' parameter. Nick. ___ Asterisk-Users mailing list [EMAIL PROTECTED]

RE: [Asterisk-Users] Call files

Gary White (Network Administrator): Nope that will not do it. That's still specifying a channel. I want to specify a number, context, and priority if there is some way possible. Then the call will follow the dial plan if I have forwarded my number to another extension. I would have

RE: [Asterisk-Users] Asterisk incoming calls

Mauro Locatelli: Is there a way to distinguish among incoming calls in asterisk. Yes. (i.e. +39-1234-JOHN goes to John's phone, +39-1234-HENRY goes to Henry's phone). OK. Is it possible? Yes. If yes, can you provide with an exten rule example? First off, make sure the line in

RE: [Asterisk-Users] Notification of missed calls

I asked: I would like some form of e-mail notification to be sent when a call is dropped before it's answered, or if it fails through to voicemail, but is dropped before a message is left. Does anybody have any idea on the best way to implement this? And there was no reply. Which

RE: [Asterisk-Users] Intercept HOLD of Snom phones

sure that this doesn't happen. Nick Barnes ___ 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] Intercept HOLD of Snom phones

,SetMusicOnHold(INTERNAL) and for external calls, insert the line: exten = whatever,whatever,SetMusicOnHold(EXTERNAL) in the appropriate places. Voila. Nick Barnes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo

RE: [Asterisk-Users] Setting CallerID on UK BRI line

or set it to an invalid number, BT will default the caller ID to the base number of the ISDN device, so remove the 'SetCallerID' line to do your testing. Otherwise, you need only four or six least significant digits of the number for the SetCallerID command. Nick Barnes

RE: [Asterisk-Users] No sound from IVR scripts, yet calls placed without any problem.

coming, there's sure to be something I missed. I think I may just recompile * and try it all from scratch again. Hmmm. Nick Barnes Senior IT Consultant. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo

RE: [Asterisk-Users] English vs American voice files

correctly. That's my 2p anyway. Nick Barnes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] No sound from IVR scripts, yet calls placed without any problem.

Nick Barnes ___ 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] No sound from IVR scripts, yet calls placed without any problem.

other than the extra HFC card (I think this is a bit of a red herring, but I mention it anyway). This box was also fully working in my office before I took it to his. Nick Barnes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com

RE: [Asterisk-Users] extensions.conf

become apparent. Nick Barnes Senior Consultant. ___ 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

RE: [Asterisk-Users] Asterisk not outputting real time display

and if there are any errors. Try starting Asterisk with -, or type 'set verbose 9'. Cheers, Nick Barnes Senior IT Consultant. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE

RE: [Asterisk-Users] Leading '0's and what do 'pri_dialplan', 'pridialplan' and 'prilocaldialplan' in zapata.conf do?

nationalprefix=0 internationalprefix=00 in your zapata.conf file! Ah-ha. Bingo. And, after some googling, I found an explanation of why this is necessary under the capi.conf documentation. Doh! Many thanks to everybody who replied on- and off-list. Your help is very much appreciated. Nick Barnes

[Asterisk-Users] Leading '0's and what do 'pri_dialplan', 'pridialplan' and 'prilocaldialplan' in zapata.conf do?

skills started to rust 15 years ago and they're almost completely non-existant now :-( Many thanks in advance of your help, Nick Barnes Asterisk version -- CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a zaptel.conf -- loadzone=uk

[Asterisk-Users] Crossed lines - a worrying problem.

LAN with three PCs and a printer, no strenuous file copying etc. Network utilisation is very low. All the kit have 'real' IP addresses. Has anybody got any ideas what could possibly be causing this worrying problem? Cheers, Nick Barnes ___ Asterisk

RE: [Asterisk-Users] Re: Crossed lines - a worrying problem.

, which would sort of make sense. Perhaps. Hmmm again. I think I may replace it with a HFC card and Zap channels and see what happens. Cheers. Nick Barnes Senior IT Consultant. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com

[Asterisk-Users] Problem with HasNewVoicemail()

/201/' definitely exists. The Asterisk version is - CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a Has anybody else seen this error or knows what stupid mistake/assumption I've made? Nick Barnes Senior IT Consultant. ___ Asterisk-Users mailing list

RE: [Asterisk-Users] Advice on BT ISDN Services (UK)

in PTP mode *and* have MSNs on it - they are completely incompatible. Like I said, you've got to love them! Ho hum. Nick Barnes Senior IT Consultant. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk

[Asterisk-Users] Astricon hotel recommendations.....?

much is it likely to cost (in $ per mile)? Is anybody else staying elsewhere and would be willing to share transport costs? Cheers, Nick Barnes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Astricon hotel recommendations.....?

over longer and depart on the Sunday, so I'll need 5 nights of accommodation, hence my need to keep it cheap - I can't afford $111 x 5. Cheers, Nick Barnes. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo

RE: [Asterisk-Users] Astricon hotel recommendations.....?

between the hotel and Brookhaven and the hotel shuttle is for hotel residents only. So... How do I get from Brookhaven to the Marriott? Nick Barnes ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Advice on BT ISDN Services (UK)

with different installation costs, rental charges and call allowances - if you don't specify which one you want, they'll pick one for you (and probably at random). Have fun, and if you want to talk off-list, please feel free. Cheers, Nick Barnes Senior IT Consultant

RE: [Asterisk-Users] zaphfc - hfc pci based ISDN card : point2point DDI

Robinson Tim-W10277: We are using the HFC card in point-to-point mode with DDI. I am using bri-stuff-0.0.2 as well. So, reading between the lines To enable DDI/DID/DOD, the card must be run under the ZAPHFC channel driver and therefore must be a HFC card? Can somebody confirm this?

RE: [Asterisk-Users] zaphfc - hfc pci based ISDN card : point2point DDI

Holger Schurig: Basically yes, but ... Many thanks for your help - I'll stop playing with the AVM cards now! HFC cards are cheap as well. Check the voip-info.org wiki, as usual :-) Indeed. Had a look there and found a few cards, but what I really was after was recommendations for a good

[Asterisk-Users] Unavailable/Withheld identification

Hi, I'm in the process of switching over to Asterisk from Alchemy kit and have hit a stumbling block. We're in the UK and use ISDN. At the moment we don't accept calls from withheld numbers (we just play them a message), but do accept calls from unavailable numbers. There doesn't seem to be any

RE: [Asterisk-Users] Kernel panic with two Fritz cards

Thomas Karcher: possible IRQ conflict? Ah, no. I finally worked out what it was... All my fault is the short answer! I was using kernel 2.6 with the SuSe 9.1 fcpci drivers. I dropped back to 2.4 and used the SuSe 8.2 drivers and it all appears to be working fine. Ho hum. Nick.

[Asterisk-Users] Kernel panic with two Fritz cards

So I've got a PBX running with one Fritz card with the fcpci module and that works fine, but add another one and... kersplat... Kernel Panic. It boots fine with two cards, but panics as soon as the first fcpci.ko module is 'modprobe'd - I don't even get as far as modprobing the second. Any

RE: [Asterisk-Users] Kernel panic with two Fritz cards

Hi, Are you sure that you compiled the module with exactly the same gcc as the kernel? Yup - I did the kernel compile, rebooted and then did the module compilation (and then asterisk and then chan_capi etc.). No other changes were made to the system. Cheers, Nick. -- BCN Information