similar to: HFC ISDN card with bristuff from junghanns.n et?

Hi Alessio Yes, the problems you report do seem similar to the issues I had. I found on the Dells that the audio prompts were very choppy and played slower than normal. Occasionally there would be 'bursts' oav a second or so of 'good' audio. I also suspected IRQ issues but the Dell Mobos had no way of adjusting them. Best thing is to try and get the card on its own unshared

hello, anyone has worknig ISDN hfc-pci card in DDI (DID) point2point mode? what kernel ? and second question mISDN driver .. anyone has working solution with mISDN and maybe fritz card? what you suggest for DDI -> point2point mode (card,kernel,chan_..., ...) ? thank you, Tomaz

Hi, has anyone succesfully installed such scenario ? I'm having problem with Award bios mb pc's... it do works with others, what's your idea ? Tnx ! -- Best regards, Alessio mailto:afoc@interconnessioni.it

Hi, I would like to use Realtime extentions with a four bri card, the classic "quodbri". Normally with that card I would use * "bristuffed" from Klaus-Peter Junghanns, but since that package is based on stable version there is no Realtime at all in it (I suppose). Any idea, other than wait for realtime to begin stable ? :) Tnx ! -- Best regards, Alessio

The update worked like a charm! Hold music is as cheesy as ever! Thanks much, this list is a life saver! Dan ------------------------------ Message: 2 Date: Fri, 18 Mar 2005 09:16:59 -0600 From: Eric Wieling <eric@fnords.org> Subject: Re: [Asterisk-Users] Redhat 9 Music on hold To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com>

Hi, I would like to enable mysql handling of voicemail boxes ... following that tutorial http://www.voip-info.org/wiki-Asterisk+voicemail+database so I modified the makefile of /apps directory to include USE_MYSQL_VM_INTERFACE=1 and copied mysql-vm-routines.h in the /apps dir, set up the db and some boxes in the table, also edited the voicemail.conf file. Everything compiles just fine, then

Hi, I'm testing realtime right now, it does not seem to me that realtime contexts can be included in normal context, like this [sip] include=>sip-dial exten=>i,1,Hangup [sip-dial] switch=>Realtime/sip-dial Am I getting it wrong ? Tnx ! -- Best regards, Alessio mailto:afoc@interconnessioni.it

Hi, I get this message after password request in voicemail app: Unable to create lock file: No such file or directory Anyone got a clue about fixing that problem ? I can't understand what directory or file we are talking about .. Tnx for any help! -- Best regards, Alessio mailto:afoc@interconnessioni.it

Hi, I'm having some problem with realtime: let's say I have a dialplan like this [globals] IPPHONES=_3XX [sip] exten=>${IPPHONES},1,Answer A call from ip phone 300 comes in, and it's been answered. Then I "switch" the sip context to realtime, putting the exten in the db and using the variable name for this as in the file version. [globals] IPPHONES=_3XX [sip]

I got this compiling the new cvs code ... any idea ? Tnx ! gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-04/07/04-11:28:50\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"

Sorry for the stupid question: What's the purpose of defining a peer as trunk in iax.conf ? The question is also valid generally speaking (for other channel types), for instance: why define a Zap group as trunk in extension.conf ? Tnx for any help ! -- Best regards, Alessio mailto:afoc@interconnessioni.it

Hi, since I run asterisk as root with a CLI open on TTY12 I was wondering if the "!" (shell) command can be disabled from the config, for safety reasons it seems me usefully. Tnx for any help ! -- Best regards, Alessio mailto:afoc@interconnessioni.it

Hi all, I have a box running asterisk with T410 connected to a Nortel DMS 100 switch and another box running SER with grandstream phones on it So if there is a call from the pstn it goes from the Nortel to the asterisk and then to the SER box and finally to the phones.if the phone is busy or the number is invalid the * box will first send an ALERT message to the Nortel and say the call is going on

Hi. I'm trying to get a Beronet QuadBRI card work with bristuff drivers. Though qozap module loads right, all card spans are in deactivated status. I'm quite sure my configuration is correct and using a single BRI card instead of the quadBRI the status is active and I can place and receive calls. On Beronet installation manual I read that Beronet and Junghanns cards are identical in

Hello everyone. I've got a HFC ISDN card that I'm using with chan_misdn and it basically behaves like crap. Echo is waaay worst then echo I get TDM400 card, sound is "choppy" (there other side is allays complaining about sound interruptions) and to top it all it detects fake DTMF's all the time. Is this a chan_misdn problem or is it a card problem? I really need to get

My suggestion would be the one-line eyeBeam phone under development. Check out support.xten.com. //Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r afoc@interconnessioni.it Skickat: den 18 oktober 2005 14:48 Till: Asterisk Users Mailing List - Non-Commercial Discussion ?mne: Re: SV: [Asterisk-Users] Queues

Hi there, I'm going mad at this: Asterisk with one HFC isdn card, using the zaptel driver "bristuff" All works ok, but voice coming in/out of the isdn card is out of sync, squelky and disrupted, UNTIL I PUT SOME LOAD TO THE PC, let say launching xwindows. I noticed this: Strong HDD activity = voice is good HDD doing nothing = voice is not good I suppose this could be an

Hi folks! Based upon your experience on the field what wifi sip phone would you reccomend ? A customer asked for a wireless * install and I'm looking for advice, tnx Alessio Focardi [[*] - Interconnessioni Italy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060628/05b2fb30/attachment.htm

HI, quick and simple question: is it possible to use inband dtmf with g729? What I would like to do is to have sip clients connected to asterisk and a zaptel card to make pstn phone calls. My concern is to allow sip users to use digits for call destinations that do require menu actions while retaining low bandwith occupation. Tnx ! -- Best regards, Alessio

Hi, I'm experimenting attended calls tranfers and I'm a little bit confused. In usual pbx's normaly there is no difference between an attended call transfer and a blind one: you just hit "transfer" then dial the extension you want the call to be transfered. If you stay on the phone you can talk to the other party, then, when you hangup, he will get the call. If you hang