similar to: DISA and repeating calls

After switching to Asterisk 1.2.14 from 1.0.x, I encountered a very strange problem. There is no sound with Playback() or Background() commands. Even though, Asterisk console shows the file is being played when I call the extension (i.e. echo test), I can't hear anything. My echo test extension looks like this: exten => 600,1,Answer exten => 600,2,Playback(demo-echotest) exten

Hello, I'm wondering how to make it possible for the user to cancel the last entered digit, if he made a mistake. For example, a user calls and starts entering 1...2...4, then he should be able to press, lets say *, to cancel 4 and enter i.e. 3. Thanks Jake -- ------- Domeny w ULTRA NISKICH cenach: ------- .pl - 29 zl, .com.pl - 22,50 zl, reg - 7,50 zl

Hello, I have two asterisk servers and I would like to merge their dialplans. I thought DUNDi would be a natural choice. I created the following configuration on the first server: iax.conf Code: [dundi] type=user dbsecret=dundi/secret context=dundi-local dundi.conf [general] ttl=4 autokill=yes cachetime=30 entityid=00:06:5B:8E:B0:08 secretpath=dundi bindaddr=XXX.XXX.XXX.XXX port=4520

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Today I have changed my DSL modem and problems with postfix begun. It receives mail, but cannot send any. In logs I can read Jun 22 19:39:53 server postfix/smtp[5040]: 9A51711FF85: to=address at somedomain.com, relay=none, delay=12840, status=deferred (Host or domain name not found. Name service error for name=somedomain.com type=MX: Host not found, try again) My resolv.conf is proper. I can

Peoplefone AG jest firm? oferuj?c? us?ugi telekomunikacyjne oparte na technologii Voice over IP (VoIP) w wyj?tkowo niskich stawkach. Peoplefone jest certyfikowanym partnerem firm Siemens<http://www.siemens.ch/index.jsp?sdc_p=c175fi1012637lmno1012637psuz1&sdc_sid=1113876080&>i AVM/FRITZ!Box <http://www.fritz-shop.ch/>. W zwi?zku z naszym dynamicznym rozwojem, do nowo tworzonej

Hi, # doveadm -Dv quota get -u pit doveadm(pit): Debug: auth input: pit system_groups_user=pit uid=500 gid=500 home=/home/pit doveadm(pit): Debug: Effective uid=500, gid=500, home=/home/pit doveadm(pit): Debug: Quota root: name=INBOX backend=fs args=mount=/var/spool/mail doveadm(pit): Debug: Quota rule: root=INBOX mailbox=* bytes=19922944 messages=0 doveadm(pit): Debug: Quota warning:

I am setting up a test domain to learn bind, and how it,s working. The master domain is bbb.ttt.domene 10.11.12.13 the child domain ftp.bbb.ttt.domene 10.11.55.77 The thing I don't understand is, why don't the query report additional info (ip address)? Her is two queries(the first the one that troubles me), cut of named.conf, test of zone file, the zone file and cut from log file.

This may be sorta off topic considering this is probably regarding the linux kernel but I am hoping someone here will have the answer anyway. I am having the strangest problem when mounting a samba share with "-t cifs" as opposed to "-t smbfs" where, instead of mounting as the user I am logged in as, it is mounted as if I were logged on as a completely different user. For

I filed this as bug 4150 http://rsync.samba.org/cgi-bin/rsync/ We had a few hundred hung processes and reached the limit for "max user processes" (see the ulimit command). rsync did not handle this very well. It passed -1 as a PID to kill(). When kill() gets -1, it kills all of the processes owned by that user, or if run by root, all processes. I attached a one line fix. Let me

Dear list, I have problems with DISA on an specific server with Asterisk 1.4.26.2. After starting DISA I can only press one key and DISA is jumping direct into the context without waiting for further digits. In dtmf.log I found this: [Nov 6 00:09:28] DTMF[2413] channel.c: DTMF begin '7' received on SIP/214-00d92db0 [Nov 6 00:09:28] DTMF[2413] channel.c: DTMF begin passthrough

Hi Friends, I am facing a strange problem with DISA. I have installed and configured Trixbox. I've created a secret extension i.e., 555 and called this extension in Digital Receptionist using custom extension i.e., created in extensions_custom.conf file. When I call from my mobile phone to my PSTN number, which is connected to FXO port, my IVR is responding. After entering my DISA

Hello, Im tryng to setup DISA on my server. Outlines comes via VoIP to my asterisk server. When i dial from outside to my disa number it answers. I dial the extension that i want to dial but Dial tone keeps up playing about 3-4 seconds more even i start to enter numbers..Then when timeout occurs, it dials the number. What is strange is even if detects the dialed number in a mysterious way, if

When I call into my Asterisk box via my VoIP line (using gsm codec) and then try to make an outgoing DISA call over PSTN I get the following: [Nov 1 15:12:54] WARNING[17694]: chan_dahdi.c:8930 dahdi_write: Cannot handle frames in gsm format [Nov 1 15:12:54] WARNING[17694]: app_dial.c:1401 wait_for_answer: Unable to forward voice or dtmf Obviously, it looks like asterisk is not converting the

Hi/ I am newbe in asterisk. I try to setup callback with Disa on my home server Anybody help me, pls

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Hi all, i use the DISA app for giving the user a trunk after a authentication through PGSQL as follows .... auth via PGSQL exten => s,1,DISA,no-password|test I think the user is now in context "test" and he could dial any number if the extension-conf in "test" is for example exten s,1,Dial,OH323/<myip> But if the user dial one digit the call build up

hi all the help output for DISA ends like below, with the half-sentence 'Note that in the case' what's the rest of that sentence? The file that contains the passcodes (if used) allows specification of either just a passcode (defaulting to the "disa" context, or passcode|context on each line of the file. The file may contain blank lines, or comments starting with