similar to: Polycom + Voicemail + Display message envelope in LCD

Hi, I have installed samba 2.2.8a for Sparc platform. I have configured it in this way [global] workgroup = DOMINIO null passwords = yes netbios name = SERVIDOR wins server = servidor_wins security = server password server = servidor_passwd name resolve order = wins bcast host lmhosts server string = Servidor Solaris Unix con

Hi, I am trying to set up an E1 line with CAS signaling using available unicall patches with libmfcr2 implementation. Inbound calls works well, I am able to get DNIS and ANI from incoming call, but I am still not able to make an outbound call with our local carrier. After tweaking of protocolvariant parameter in unicall.conf I was able to find proper values for inbound and outbound calls, but

The bar is getting raised yet again http://www.voipmonitor.net/2007/06/05/New+Features+And+Services+For+Pack et8+Virtual+Office.aspx I personally use Snapanumber $30 or there abouts (after trialing a few other TAPI solutions and finding them sub-par) and think it's a great product but interesting to see how more people are expecting desktop/phone integration applications. Does anyone

Hello, We use MS Access 2000 (I know, we're migrating away from it) as an application to automatically dial phone numbers. The old phone system we have allowed the call representative using the application to take their phone off hook, push a button in the app, and the app would send the phone number to the phone system and dial the number. We are moving to Asterisk for our main phone

Hey folks, So I'm trying to get Festival() working on 1.2.17. I'm trying to use app_festival: Here's the show dialplan output from that extension: '3378' => 1. Answer() [pbx_config] 2. Festival(Hello Asterisk caller. How is your day?) [pbx_config] 3. Playback(vm-goodbye) [pbx_config] 4. Hangup()

All, I've a question on A*k hardware. I'm running 1.2.18 on a Dell DC051 (Intel(R) Celeron(R) CPU 2.80GHz) with 512mb RAM. I'm supporting 60 users (Cisco 7940s each + Xlite PCs). Call loads are low, max of about 10 simultaneous SIP/IAX calls. CPU for A*k rarely goes above 2% as I can tell. Its IP only, no E1/T1 cards present. However, I get complaints of bad voice quality,

Hi, I'm trying to find a way of passing the actual call duration (something like ANSWEREDTIME) to an AGI script that runs periodically during a call. Any ideas? Thanks, Adi. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070531/115b0aee/attachment.htm

Hi, I just got a Polycom 330 and, of course, I don't have the firmware and sip.cfg files to provision it. Where can I get those? Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070605/a14615c8/attachment.htm

Hi all, Does anybody know why my asterisk doesn't have a "show agi" command? Do I have to load any module for it? Thanks Ronaldo.

Dear Users ! I have recently installed asterisk 1.4 i got a warning message whenever i use reload or extensions reload. [Jun 29 19:22:11] WARNING[4539]: pbx.c:6236 ast_context_verify_includes: Context 'ael-local' tries includes nonexistent context 'ael-parkedcalls' [Jun 29 19:22:11] WARNING[4539]: pbx.c:6236 ast_context_verify_includes: Context 'ael-dundi-e164-local'

I am trying to get updateconfig working. I found an example of updating configuration files here: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Upd ateConfig When I tried it the conf file was updated but the new entry was not added. action:updateconfig reload:no srcfilename:manager.conf dstfilename:manager.conf Action-000000:append Cat-000000:newuser

I am working on a call-back solution where the initiating call should never be answered. I was doing this simply through the dial plan, sending a progress tone, and then dumping the channel, and firing off a DeadAGI which created a call file to make the callback. Now I've tried extending this so that an AGI is fired first to check for things - like no inbound ANI - and play a

I'd like to be able to generate a Manager Event from the dialplan but can't seem to find a way to do it. Alternatively, trigger and Event when a record in AstDB gets changed. Can anyone point me in the right direction? Thanks. By way of explanation, I've a app that connects to astmanproxy and I'd like it to know when a call group gets put into Nightservice. Putting the call

Hello there, I appreciate any help about this problem that I can't figure out... I need to record all my calls: this is pretty easy using Monitor() before the Dial(). eg: exten => 425,n,Monitor(wav49,/var/spool/asterisk/monitor/425/${EPOCH}_${CALLERID(num)}_in,mb) exten => 425,n,Dial(${PHONE1},10) Now, I want to create a call group: I mean, I want a number (eg 800) that makes

Hi All Just been looking at stats for one of my sites, and I'm conserned about the number of error cause codes being returned from the telco for example 12000 calls processed 131 are cause code 31* normal. unspecified.* 139 are cause code 28 * invalid number format (address incomplete).* 112 are cause code 1 *Unallocated (unassigned) number. *this adds up to about 3% of calls not

Hi, Lately we've noticed that some Zap channels on one of our PRIs are unavailable. We have 2 PRI lines with 60 channels in total. On the first PRI there are currently 20 channels that are not being used for some reason. I tried googling around and found some similar problems but there really was no solution (?). I'm not sure if this problem has occured now because of more load on the

Hi, I am running dovecot version 1.2.9 on a Ubuntu 10.04 server. To obtain the message's header info from this dovecot imap server, I run the following imap command on a client side's imap session: . fetch 88:90 (envelope) which returns (in this case) ------------------------------------------------------------- * 88 FETCH (ENVELOPE ("Tue, 28 Oct 2008 15:45:35 -0500"

Con parse_number sacaria todos los numeros de la columna, pero lo que busco es sacar solo los que empiezan por AV y descartar el resto. De todas maneras muchas gracias, le voy a dar otra vuelta. ________________________________ De: Juan Carlos Lopez Mesa <jclopez5 en unal.edu.co> Enviado: martes, 22 de septiembre de 2020 21:32 Para: Samura . <tontito82 en hotmail.com> Cc: r-help-es

I know that I can record the contents of a call by calling Monitor() or MixMonitor() from the dialplan just before invoking Dial(). I have a potential customer who wants only the first minute of each call recorded (for identification purposes, without the storage overhead of keeping the complete call). Can anyone here think of the easiest way to do this? The only possibilities I can think of

Hi all, I have four servers (one with redhat 7.3, two with CentOS 3.4 and one with CentOS 4 for testing). I would like to do a partial package mirror with os, updates, contrib branches and so on. Anybody knows if exists any tool for redhat based distros, like apt-proxy in debian world ( that donwloads packages on demmand), that supports this feature?? I see yam tool from dag repository, but it