similar to: REALTIME and VARIABLES

Hi, I tried set up a global var for an extension, like this [globals] IPPHONES=_3XX [sip] exten=>${IPPHONES},1,Answer What I would like to do is to make a dialplan without fixed extension numbers to change the entire dialplan according to the customer requests: what exten number do you want for your IP Phones ? let me change a variable and we are set! It seems that this is not supported,

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

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\"

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

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, 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

Please can you explain in more details as to what your problem is? I have 2 cards working in one PC, but have had problems with Dell motherboards. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Alessio Focardi Sent: 17 June 2004 11:41 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] HFC ISDN card

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, 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

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

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

Hello *'s, I have question regarding Quintum Tenor AXT800 VOIP gateway can anyone integrate it with asterisk if anyone what is the scenerio i have scenerio which is quite simple but i am confused about it whether it is possible or not : I integrate it with asterisk for interanet no PSTN at all just only IPphones and analog phones connected on FXS port.Is it's neccassary to cannect with

Hi, This issue has been discussed probably a million times on every asterisk forum in the world and I have the same problem too. Another problem you would have with the agents is that when they make an outgoing call they are not regarded as "busy" by asterisk and it sends more calls to the agent if it has call waiting enabled. This behaviour is totally senseless since the whole purouse

Hi, I'm running 1.2 beta1 in a mini call center. I have 3 queues with 10 operators, and I'm running into some trouble because when all the operators are busy answering call asterisk still sends them more, resulting in a "beep beep" (call waiting) over and over again in Xlite audio. An easy solution woud be the use of a "single line" user agent, like firefly, still

I am planning to sign up for a VoIP service in the U.S. Can anyone recommend anything cheap, reliable and good quality? I want to use it for my primary house phone (I also own a cell phone). I also want the service to be asterisk friendly so I can play with it :-) Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am unable to get a my Draytek working with our Asterisk server. I can make/recieve calls but get no audio. I have tried the various codecs at the Vigor end but still getting nothing. I looked at sip debug (below) but am new to Asterisk and don't really know what I am looking for. Asterisk works fine with XLITE so I know my installation is ok. Sip read: INVITE

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

Hi I have installed the http://asteriskathome.sourceforge.net/ with a Digium card with no problems, very good ! Now I want to install my Billion PCI ISDN card (HFC based) in TE mode. I get a little confused with Isdn4Linux, ZapHFC HIAX and the need to install Capi ! Please suggest best and easiest approach ? Thank you ! HB Norway