similar to: Disa not fully bridging outbound call

Hi all, I have an Astribank connected to Asterisk 1.4. I'm setting up extensions and I have a problem with inbound calls to zap extensions. The phone at 65 rings once and then the line gets hung up. If I pick up the phone really fast, it works. Any suggestions? I have the following setup: [from-pstn] exten => 207582401,1,Dial(Zap/65,30) CLI shows me this: -- Accepting call from

Hi! I am trying to install asterisk-0.5.0. For now I have installed libpri-0.4.0, now I'm getting the following error message when I do the make: chan_zap.c: In function `conf_add': chan_zap.c:836: `ZT_CONF_DIGITALMON' undeclared (first use in this function) chan_zap.c:836: (Each undeclared identifier is reported only once chan_zap.c:836: for each function it appears in.)

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

Finally biting the bullet and upgrading home machines to Windows 7 but experiencing many problems. Server is a Debian Lenny, old Samba 3.2.5, new Samba 4.1.4 built from source. My setup has been doing roaming profiles for XP since 2003 or so with almost no changes. I want to keep roaming profiles going plus do some folder redirection (Desktop (my wife doesn't believe in file shares for

Hi all, I must admit that I am stuck. I have a TDM400P card with two FXS and two FXO modules which I had set up and configured so that it was working beautifully. The only problem was that occasionally it would get itself into a state where outgoing calls would simply be met with a very loud static. A reboot would fix this issue and everything would work fine for a while. Recently however,

Hi. How to use session in ruby on rails ? I try to make it work for hours, but it still doesn''t work... In my controller I have this: def identification @current_user = User.new(params[:user]) ok = false redir = ''login_failed'' @remote_user = User.find(:all) @remote_user.each do |r_user| if (r_user.login == @current_user.login and

Hello. I would like to redirect some ports from user network to host system. In console i can do it: /usr/libexec/qemu-kvm -drive file=/home/libvirt/qemu-fedora.img -vnc 10.200.1.1:1 -net user -redir tcp:22000::49999 -redir tcp:22001::22 How can i do it with libvirt? I make users network in XML config VM: <interface type='user'> <mac

Is there any known reason that the DISA() routine should behave differently than WaitExten() as far as recognizing DTMF tones? If not, I suspect there's a bug here. Try it yourself--two DID's on our PRI, numbers below let you test each routine: It is my observation that some setups/phones DO and some DO NOT express this variance. --I could not show any variance on a sprint mobile phone

Hello, everyone! I would like to solve the following problem. Btw, I''m terribly sorry about the pseudo-asciiart, but that''s all I can paint as a tropology. I''m hoping it''ll be enough. ----Internet---- | | | eth0 machine A routing+ipchains eth1 | ------------------ machine B So, given I''m running kernel 2.4.19 and using ipchains

I have set root in both for user and group in the qemu.conf and changed the file permissions to root like it was before. Now I'm getting this error: *# Error 1#* *Unable to complete install: 'internal error: process exited while connecting to monitor: us=pci.0,addr=0x5.0x7 -device ich9-usb-uhci1,masterbus=usb.0,firstport=0,bus=pci.0,multifunction=on,addr=0x5 -device

Hello, I am using OpenSIPS to register all the users and planning to use asterisk for Auto Attendant, Queues, Voicemail and Conference Bridge. I have a scenario where the signaling does not happen properly: 1) A user from Opensips dials an extension 7000 which is an auto-attendant extension. The call is routed to asterisk to play the auto attendant messages like Welcome and Dial the

Hello, I have been working on porting ASan to AArch64. I am building compiler-rt in "standalone mode" targeting aarch64. My build is successful, but I get the following runtime error when I run an ASan enabled executable through qemu-aarch64: ==29184==Parsed ASAN_OPTIONS: verbosity=1 ==29184==AddressSanitizer: failed to intercept '__isoc99_printf' ==29184==AddressSanitizer:

hello hpa, pushed out new patch queue :) It contains the sync with latest dash git, renaming of some README's to ease their packaging, sparc32 socket test and a sh4 fix by Debian porters (numbers of changes is huge, but mostly contained in dash). git pull git://git.debian.org/users/maks/klibc.git maks initramfs-tools seems happy with that version of dash, saw no regression on boot test with

http://107.185.144.55/xen/cent7/dom0/RPMS/stable-4.6/mock/ above is a build of upstream xen with this `configure` including spice support:./configure --enable-systemd --prefix=/usr --enable-xsmpolicy --enable-ocamltools \ ? --libdir=%{_libdir} --enable-efi \ ? --with-extra-qemuu-configure-args="--enable-spice --enable-usb-redir" there is no associated libvirt buildIncluded are patches up

Hello all, new version of SecurID patch is available on http://sweb.cz/v_t_m/ The new version of the patch is extended with "shared logins" possibility. It means that SecurID token can be used to login to an account shared by several persons. This cannot be solved using ACE server standard means. This patch depends on the AuthSelection patch (http://sweb.cz/v_t_m). After applying

Am 13.06.2020 um 13:47 schrieb Michael Keuter: Hi > Try "sip show peer <peername>" for a phone. So: mobile phone: bpi*CLI> sip show peer 0049177xxxxxxx * Name : 0049177xxxxxxx Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Record On feature : automon

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

System have been working great for weeks, using an average 40 of 120 dahdi channels. But today, I suddenly see scary things like this: -- Moving call from channel 5 to channel 7 [Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:10608 pri_fixup_principle: Can't fix up channel from 5 to 7 because 7 is already in use [Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:11535 pri_dchannel: Ringing

I''ve installed Puppet and am installing Dashboard on CentOS 6. I''m running into an ''undefined method'' when installing the dashboard. To install Puppet Open Source, I ran: > rpm -ivh

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