similar to: SIP call logging, called number not logged

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

I got this today trying to place a call through FWD: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 217.11.11.1:5060;branch=z9hG4bK230f856c From: "Iain" <sip:12345@fwd.pulver.com>;tag=as6eaa85fb To: <sip:10001@fwd.pulver.com>;tag=b27e1a1d33761e85846fc98f5f3a7e58.3701 I didn't used to have any trouble with FWD and * is registering with FWD OK. Has

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

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

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

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

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

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

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 =>

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

I've been tinkering with ENUM and found that the lack of a debug message in enum.c that says it has actually succeeded in resolving an address is a bit of a nuisance. It makes it difficult to see if failures with ENUM are due to problems with parsing NAPTR records (in enum.c) or mistakes in extensions.conf An extra line of debug information would be much appreciated! Iain

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

Does anyone configure that or is that possible ? Thanks in advance -- Best regards Vlad

Hi, Does anyone know where I can get a telephone card that will fit into the PCI slot on my PC and work with the UK telephone system (BT) ? I would really like the retailer to be based in the UK if at all possible ? Also, is there any way to set up asterisk so that only certain phones can make external calls, but all phones can receive incoming calls if they are routed to that phones by some

Hello-- I'd like to do a little processing on external phone numbers from within the asterisk pbx. Fairly simple stuff, but... devilishly hard to make it work so far! 1. I'd like to dial 9 to get an outside line. 2. If the number dialed after the 9 is 754XXXX, I'd like it to go thru unmodified. It's the only local number available here. 3. I'd like all 1 XXX XXX XXXX numbers

Hi all. I'm looking to set up a fax via email service so that users can email a specific mailbox and receive fax's to a specific mailbox. Can this be done? I've had a look an SpanDSP and I think that's what I want but I'm not sure. Cheers Matt

Sent this one in the other day but did not get a reply! When first time logging into Samba (or share), using an dialup internet modem connection, it seems to take between 50-70 seconds to respond to a Windows 98 request. You can see the request leaving the modem and can count the 50-70 seconds before the modem responds from the Linux Server box. Don't see the same problem if I telnet, ftp or

Hi, I''ve managed to create a web based interface to some of the advanced routing capabilities in Linux. Currently it is functional (and pretty, i think :) although far from perfect. I''m looking for people with better programming skills and/or understandings of advanced routing concepts in Linux than I have who''d like to help out with the project. The overall goals