similar to: CLI variable not set on incoming call

Hi, need a quick help ... it should be easy but ... exten =>_9898,1,Answer exten =>_9898,2,VoiceMailMain(${CALLERID}@domain) Accepting overlap call from '342' to '9' on channel 0/2, span 3 -- Executing Answer("Zap/8-1", "") in new stack -- Executing VoiceMailMain("Zap/8-1", "@domain") in new stack As you can see there

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 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 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

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

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]

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

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 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, 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 would like to know if there is a way to detect the status of a span prior of sendig a call across it from the dialplan. I was asked to set up an * server with 2 spans connected to the telco and use the second as failover for the first. I checked that dialing a failed span (for instance: cable disconnected or no line) results in "congestion" for ${DIALSTATUS}, but message is too

Here is the error I get compiling the asterisk-addons rpm cc -fPIC -I../asterisk -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: error: `PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP' undeclared here (not in a function) make: *** [cdr_addon_mysql.o] Error 1 I'm pretty sure I got all that is needed: # rpm -qa | grep mysql mysql-3.23.58-4

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'm looking for an hardware platform for an * installation that should have at least 3 PCI slot with no irq sharing whatsoever. Hardware raid 1 with hot swap is a premium, but not mandatory ... What would you choose? compaq/hp ? Dell ? Ibm ? Tnx for any advice on this matter! -- I migliori saluti, Scrivi a: Alessio afoc@interconnessioni.it

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

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

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\"