similar to: FWD trouble - 407 error

A caller to me was this afternoon detected as a fax machine: Jan 15 15:31:17 NOTICE[41997]: File chan_zap.c, Line 3564 (zt_read): Fax detected, but no fax extension ... and then redirected to voicemail. An extract from extensions.conf is attached below. Is there any way to stop * even considering an incoming call on a line as a fax call? Iain bell] include => mailboxes include

I've just noticed that if you start a call by writing a file to /var/spool/asterisk/outgoing the cdr created on termination logs the call placed to the local extension - not to the destination in the PSTN. Hence there is no record of the PSTN number dialled. I guess most people want to log the outgoing portion not the local call leg? Anyone know of a setting that changes this? Iain

Hi list, I've got Samba 2.0.6 running on Yellow Dog Linux with a 2.2.14 kernel. When I start the PC, it displays the login screen (I have 3 user profiles) and I enter the username and login domain (ie the one operated by Samba). I get an error message stating that the domain login server can't be found. If I then cancel the login, go to the start menu and log off, then login there is

Hi Everyone ;) I have posted something like this before but yeilded no solid help as of yet. I am new to * and havent even setup a box for it yet as to I have no clue what I should go ahead and buy before wasting a few $k. Im looking to setup * for my office with outbound calling only with some call agents, and also remote agents so they can work from home. At this time im not looking to

This is what I'm doing it gets you both sides of the phone call...small size...and playable on windows through a share. My notes: On redhat 9 I have to run the following command for asterisk to start LD_ASSUME_KERNEL=2.4.1 asterisk -vvvvgc [macro-record-on] exten => s,1,SetVar(CALLFILENAME=${TIMESTAMP}-${ARG2}-${ARG1}) exten => s,2,Monitor(wav,${CALLFILENAME}) ;exten =>

I know that Asterisk supports DID, but does anyone have documentation on how to write the configuration for it? I'll be trying to setup a hybrid system where some incoming numbers will be DID enabled and others won't, so I'll need to be able to sort between the two, i.e. directly connect the DID dialed numbers and route the others to an autoattendant for extension dialing.

I've been having issues getting FWD to work. I posted this same Q to the FWD forum (no responses yet), but I was hoping someone here had some insight: http://yabb.pulver.com/cgi-bin/yabb/YaBB.cgi?board=news;action=display;num=1072263468;start=0#0 I just signed up for an FWD account (I know I had one before, but I lost it.. :) I've got it running through Asterisk - all working fine

I've just had the most appalling performance from * ever. Dialling: Cisco 7960 => asterisk => IAX produces sound drop outs so extreme that the call is useless. I noted this in an earlier post. Dialling: Cisco ATA186 => asterisk => IAX is fine. Frankly, I think this is such a bad problem that it should be sorted in advance of any of the new features that seem to be

I've just noticed that since swapping from the direct mysql cdr driver to cdr_odbc, the call duration (and anything else that's an integer) isn't logged - anyone else seen this and know the reason. The cdr_odbc driver gives no error messages and records any string based fields correctly. Iain

Hi, as I have learned from the discussion about which SSD to use as ZIL drives, I stumbled across this article, that discusses short stroking for increasing IOPs on SAS and SATA drives: http://www.tomshardware.com/reviews/short-stroking-hdd,2157.html Now, I am wondering if using a mirror of such 15k SAS drives would be a good-enough fit for a ZIL on a zpool that is mainly used for file

This seems like a piece members of this list would find interesting... === There is growing concern over the interaction of VoIP systems with the legacy PSTN, and the transmission of caller identity data--most notably, Caller ID on the PSTN. It is not always possible, or obvious how, to handle Caller ID data when moving to or from VoIP and the PSTN networks. There are even business models

====================================================== "I think about food literally all day every day. It's a thing." Taylor Swift ====================================================== Release Announcements --------------------- This is the latest stable release of the Samba 4.6 release series. Changes since 4.6.2: -------------------- o Michael Adam

====================================================== "I think about food literally all day every day. It's a thing." Taylor Swift ====================================================== Release Announcements --------------------- This is the latest stable release of the Samba 4.6 release series. Changes since 4.6.2: -------------------- o Michael Adam

I've set up * as a gateway to Free World Dialup. The called number appears not to be logged either in the Master.csv file or to MySQL - do I need to set an option? Iain

If you make an outgoing call to a conference bridge (or anything else that needs DTMF '#') then you can't use the asterisk 'T' transfer option because that is triggered by the '#" also. Is there already a solution in # for this eg use two keys to trigger a transfer rather than just the '#'? Iain

Here's a patch that changes the behaviour of # transfers in asterisk. A single # is transferred to the remote phone/system. Two # in quick succession will trigger a transfer. This is very useful for users who have basic analogue phones connected to an ATA 186. For example, when calling a remote conference or IVR system you often want a single # to be sent to the remote system - not to

Hello, I was using the doublehash.patch that Iain Stevenson had created back in August to change the transfer key from a single hash "#" to a double-hash "#". It always patches properly, but when I went from CVS 2004-01-12 to Asterisk 0.7.1 it doesn't seem to work anymore. I've attached the patch to this email and I use the following command to patch it: patch -p1

Listening to the options on the voicemail system it seems to be missing a feature for users to turn voicemail off completely. This seems a rather glaring omission. Does the feature of turning off message recording via the phone exist - or does it need a patch? Iain

Hi Folks, Recent installation with a simple TDM11B (one FXO, one FXS) that I've set up (at home). I am receiving callerID fine from the telco, as it shows up in my call detail records, AND on 2 SIP phones. However, I'm not reliably receiving it (that is, very seldom does it come through) on the analog phones. Any ideas on where to check configurations, etc? I haven't encountered this

My company, which is a mac-heavy shop in the printing industry, needed to migrate to a faster file server. As our directory trees are very large, both Samba, and Netatalk were bogging down badly on our Linux server (Samba, due to heavy CPU usage during directory listings - the case-sensitive file system issue, and netatalk because the cnid db was getting too big). Our solution was to switch to a