similar to: Infuriating problems: no dial tone, dropped calls, no voice: 1.2.13 and 1.4.11

Ok.. will be there... -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk-users-request at lists.digium.com Sent: Thursday, October 04, 2007 12:50 PM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 39, Issue 12 Send asterisk-users mailing list submissions to

Any way to get the Asterisk to do the "broken dial tone" on phones that have voice mail in their mailbox? Similar to the phone company's built-in voice mail - awh

I'm trying to compile latest zaptel-1.2.13 and I'm getting following errors: /usr/src/Asterisk-1.2.14/zaptel-1.2.13/xpp/xbus-core.c: In function ?debugfs_open?: /usr/src/Asterisk-1.2.14/zaptel-1.2.13/xpp/xbus-core.c:171: error: ?struct inode? has no member named ?i_private? make[5]: *** [/usr/src/Asterisk-1.2.14/zaptel-1.2.13/xpp/xbus-core.o] Error 1 make[4]: ***

Is it possible to make Asterisk jump into action and play a sound file as soon as a handset is lifted, instead of providing a dialling tone and waiting for the user to dial an extension? -- AJS Answers come *after* questions.

Hi, I fresh installed Fedora Core 6. I downloaded and untar the 'asterisk-1.2.13' into /usr/src/asterisk-1.2.13. When I run ' make' I get: ... ... chan_phone.c:41:29: error: linux/compiler.h: No such file or directory make[1]: *** [chan_phone.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-1.2.13/channels' make: *** [subdirs] Error 1 [root@sss asterisk-1.2.13]#

Puppet Dashboard 1.2.13 is a maintenance and bugfix release of Puppet Dashboard. This release is available for download at: https://downloads.puppetlabs.com/dashboard/puppet-dashboard-1.2.13.tar.gz Debian packages are available at https://apt.puppetlabs.com RPM packages are available at https://yum.puppetlabs.com See the Verifying Puppet Download section at:

as my first project with *, i would like to replace our old neax2400(sds) with an * server. i've got an X100p and a TDM400 on hand already. for the CO lines, the X100p works ok with fxsks signaling though there are still strange things happening every now and again but more testing is on the way. my real big problem is the DID lines which our telcos call DDI lines; (incoming calls only) i

Hello, I have an Adit 600 (3 FXS cards) hooked up to a digium T1 card in my asterisk box. I 'connected' the slots to the a:1 T1 interfaces via the command line. The slots (3 fxs) are configured with 'ls' signaling. I configured the T1 card with the same line settings as the T1 interfaces on the adit and I get green lights on both the T1 card and the T1 interface on the adit (so

Hi Everyone... I am running Asterisk 1.2.13 on Debian "Etch". I installed it from the package. I also installed the web voice mail package, which installed Apache2 and a bunch of other stuff. When I point my browser at my PBX machine, the web page says "It Works!" but of course it does not. It does not seem that Apache is configured to run the vmail.cgi script. In the

Dear people, is it possible to have presence using Asterisk 1.2.13 / SIP with Linux/Debian Etch??? I'd like to see if my intranet contacts are available, busy, disconnected.... Thanks a lot Alejandro

Is there a way to initiate a transfer using an analog handset? For instance I'm looking for a way to do something like the following: External call comes in and is answered by user A. After talking to the caller they determine that the caller really needs to speak to user B. Is there any way for user A to initiate a transfer to user B, using only their analog handset? Now to make

Hey guys, I've had a working Asterisk setup going for a while now, but am having problems with the Zhone Zplex 10b thinking that during ringing, an extension has answered the call when infact it hasn't. This only seems to happen on some of the ports and doesn't appear to be specific to the handset. Does anyone have any suggestions as to how I could go about fixing this (aside from

Hi List... I have an asterisk box with one TDM2400, it has 8 FXS's and 12 FXO's, and it also has the echo canceller... I'm running Asterisk 1.2.13 and Zaptel 1.2.12 with the Kernel 2.6.9-34.0.2.EL I'm using Polycom's 501 with the SIP 1.6.2.0041 The problem is when someone dials to or from the PSTN through the TDM2400, the voice quality is crappy...Instead of hearing:

All, I have upgraded by home machine from Fedora Core 5 (FC5) to the recent FC6 and am struggling to build Zaptel-1.2.10 and Asterisk-1.2.13 on the box... which is an Intep P4 2.8GHz HT processor box with 845 chipset, hence the kernel installed is 2.6.18-1.2798.fc6-i686 so we hve this: [root@gate zaptel-1.2.10]# uname -a Linux gate.tubby.org 2.6.18-1.2798.fc6 #1 SMP Mon Oct 16 14:54:20

I have noticed that Asterisk (version 1.2.13) is not hanging up a call when the wifi handset moves out of range. My setup is Nokia E61 connected to wifi access point (private IP range) and then to server on internet (public IP). I have been testing using the talking clock application, and walking out of range does not hang up the call. The call will continue for hours even though the handset

Hello I am trying to build libvirt 1.2.13 (latest) from source on a Ubuntu 14.04 64 bit box. After installing all the dependencies (libyajl, libdevmapper, libpciaccess, libnl), I could finish the build and install. However, invoking libvirtd throws this: root@ubuntu:/home/hvishwanath/Downloads/libvirt-1.2.13# libvirtd libvirtd: /usr/lib/libvirt-qemu.so.0: version `LIBVIRT_QEMU_1.2.3' not

Hi All; I just need to know the openion about Grandstream phone, actually I tried Budge Tone 201 and I chocked that there is a noise in the handset (zzzzzzzzzzzzzzzzzzzzzzzzzzz) always, but in the speaker the sound is good and no noise. Anyone has idea about Grandstream, and if they have a lot of problems and such noise in handset? Or my luck was bad that this phone is defected? Regards Bilal

No matter what I do, ${ANSWEREDTIME} is always 0, even on the most simplest dial plan such as: Using Asterisk 1.2.13 exten => 77,1,Answer exten => 77,2,Playback(custom/dax/S300) ; one minute file exten => 77,3,Noop(${ANSWEREDTIME}) exten => 77,4,Hangup -- Executing Answer("SIP/5402-b7b45f58", "") in new stack -- Executing

Seems to have struck a small problem.. Using a t100p & Zhone channel bank.... one extension ringing another.....the following will appear WARNING[18448]: File chan_zap.c, Line 2685 (zt_handle_event): Didn't finish Caller-ID spill. Cancelling. if we are using defaultzone=au change it to us and the problem goes away..... any possible solutions ?? Gary .

While I'm asking about the Polycom ip500, the answers for all phones where mic/handset/headset levels are adjustable would be of interest to many I'm sure. For the ip500, the default value for the handset seems to be voice.gain.tx.analog.handset="3" I've noticed that echo all but goes away when one reduces the mic volume on almost any phone. My question is, for you users