similar to: Music On Hold - not working for me...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Ddoes it feasible with * to capture a call? when arrives a call, floor bells ring and everyone can hear them in the company, then everyone can answer 'capturing' the call m. - -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 IAXTel: (700) 350-1234 -----BEGIN PGP SIGNATURE----- Version:

Hi: Did anyone use apt-get install zaptel successfully? After apt-get instal zaptel, use "modprobe zaptel", get a "FATAL modul zaptel not found". Thanks. Yan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040722/4608e1e5/attachment.htm

> > I'm running an live 128kbit mp3-upstream (icecast v 2.2.0 > on freebsd) from a > > site where the internet-connection is rather bad and drops > every ~20min for > > few seconds (I'm using streamtranscoder by the way). my > question is if > > there's any parameter to increase the buffer of the > upstreaming-source. the > > config-file of

Does the wins libnss module not "work" yet, or am I misconfiguring it ? Everything else in the whole samba/winbind realm works, I can log in as a domain user using pam_winbind, smbclient can resolve netbios names from WINS, etc etc. but 'ping NETBIOS_NAME' doesn't resolve. Here's some background data: Samba 2.2.8-0.1 on debian from master.samba.org apt repos. wbinfo -N

I'm having hard times with paging intercom Heres my dialplan exten => 777,1,Goto(intercom,777,1) [intercom] exten => 777,1,SIPAddHeader(Call-Info: <sip:192.168.16.105>\;answer-after=0) exten => 777,2,Page(Local/308 at page& Local/309 at page& Local/310 at page) [page] ; Paging context exten => _X.,1,Macro(page,SIP/${EXTEN}) [macro-page] ;

Hello Instead of using PrivacyManager, I'd rather use my own dialplan to prompt the user for a ten-digit number if they called while blocking CID. This code does prompt the user, but 1) hangs up if the user didn't type the ten digits before the timeout 2) if the user did type the right number of digits, it still hangs up instead of Returning and then jumping forth to the "cid"

I have followed all the install note for A2billing and have everything installed and configured and my asterisk works except the callingcard application. Added the following [callingcard] ; CallingCard application exten => 777,1,Answer exten => 777,2,Wait,2 exten => 777,3,DeadAGI,a2billing.php exten => 777,4,Wait,2 exten => 777,5,Hangup I am using 777 as the calling card

Hello folks, I want to write an AGI script doing this: 1-user call a number. 2-asterisk call the agi script 3-the script dial the peer 4-if the call is answered, let the call up for 1min 5-then the script hangs up the channel. I tried either in php or in java but no success. In java i did this: ////////////// exec("Dial", "IAX2/400"); boolean t=true; while(t){

After searching around, I've been unable to to find any relevant info on this. Perhaps the group can help? I am seeing something strange with a new Sipura SPA-3000 (and I've noticed this also with an IAX softphone): When I dial 777, this dialplan (in extensions.conf) is run: exten => 777,1,Dial(Zap/1/2345678) exten => 777,n,Hangup The number is answered by the called

Hello, Does asterisk supports VXML? Couldn't find much resource on that on google and wiki. Thanks Foong

Hi, I'm using Asterisk 1.2.1 on Sarge. it seems like if I call a phone and nobody answers, asterisk does not jump to the next priority...it freezes. Take a look at this: exten => 777,1,NoOp(before) exten => 777,2,Dial(SIP/7|60|g) exten => 777,3,NoOp(after) priority 3 is never executed but this worked with Asterisk 1.0.7!!! TIA Giorgio Incantalupo

Hi, I have the problem that WaitExten is not responding to key presses. Here are the sections from my extensions.conf: [globals] incoming_call=0 menu=0 announce=0 [internal] exten => 777,1,Goto(hotline,${EXTEN},1) [hotline] exten => _X.,1,Set(CALLERID(name)=Hotline) exten => _X.,n,Set(original_extension=${EXTEN}) exten => _X.,n,GotoIf($[${announce}=1]?4:10) exten =>

Problem as follows: [default] exten => 777,1,Gosub(sub,1,1) exten => 777,n,Hangup() exten => h,1,NoOp(hung up in 'default' context) [sub] exten => 1,1,NoOp(in sub) exten => 1,n,Playback(tt-monkeys) exten => 1,n,Return() exten => h,1,NoOp(hung up in 'sub' context) This works fine if the caller listens to all the 'tt-monkeys' and let's the system

Dear ALL I have audiocode MP -124 with configure in asterisk Endpoint configuration means every analog phone register in asterisk now thing is that i have one more SIP trunk with mediant 2000 [auodiocode-mp-124]-----[ * ]------[mediant 2000]-----E1 When i call from audiocode MP -124 phone i got this error -- Executing Dial("SIP/20-0889c4d8", "SIP/mediant/1")

Hi, I would like to get the status of asterisk extension with my php program. *My program as follows,* <html> <!--<meta http-equiv="refresh" content="1" />--> <?php $fp = fsockopen("xxx.xxx.xxx.xxx", 5038, $errno, $errstr, 30); if (!$fp) { echo "$errstr ($errno)<br />\n"; } else { $out = "Action: Login\r\n"; $out

Hello, I try to append a URI to the SIP dial syntax, however the URI were not shown in the sip debug message. I have read one of the post in the list which actualy show the URI string in the debug message (at the To: field). Is there any setting I need to set or turn on during compilation of asterisk? I have the head version of asterisk and my extension.conf setting is proveded below: exten

Hi I've configured my asterisk and voicemail all works fine but I want to restrict call time to voicemail that is when user calls voicemail he can use voicemail system only for a max of 5 min that is after five minutes asterisk should disconnect the call. thanks Arun -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello I noticed something nasty with the Record() function: If the user either hangs up during the prompt (ie. doesn't leave a message at all), or does leave a message but forgets to hit the # key at the end... Asterisk stops right there, so the rest of the script doesn't run: ======== [internal] exten => 777,1,Playback(leave_msg) exten => 777,n,Record(/tmp/test.wav,3,30)

Hello Some of our customers call with CID blocked. I'd like to save those numbers into a SQLite database using a command-line PHP script, so that I can... 1. Edit the CID name through a PHP web script which will just list all the customers in the database who have a phone number but no CID name set 2. Look up those customers' e-mail address from this database, and send them an e-mail

Hi, I am using Group and Group_Count to limit the number of calls to go out over a single peer as our channels with that peer is limited to 8. If we dial and outside number over this peer and then transfer the call into a MeetMe conference the Group gets decremented when it should not? This is most likely an error on my behalf, however I am not sure what the correct solution is. I have set