search for: iainstevenson

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 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

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

...but it must be first person. For this reason, > > I do not let asterisk record everything, because my employees must > > themselves determine what they're going to record. > > > > > > ----- Original Message ----- > > From: "Iain Stevenson" <iain@iainstevenson.com> > > To: <asterisk-users@lists.digium.com> > > Sent: Sunday, January 04, 2004 12:51 PM > > Subject: Re: [Asterisk-Users] help - recording both sides of a conversation > > > > > > > > > > * always records both sides of the conversation -...

...(Brian Cuthie) 7. RE: No ringing tone with IAXY (and other bits and bobs) (Rich Adamson) 8. Extensions and Include (Kevin ) 9. RE: No ringing tone with IAXY (and other bits and bobs) (Brian Cuthie) --__--__-- Message: 1 Date: Sat, 10 Apr 2004 10:53:15 +0100 From: Iain Stevenson <iain@iainstevenson.com> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: Analogue telephone cards for the UK Reply-To: asterisk-users@lists.digium.com --On Saturday, April 10, 2004 10:42:26 +0100 Paul Tyreman <paul@tyreman.org.uk> wrote: > > Thanks for all the replies. > &g...

I have asterisk set up to ring a local SIP phone (on an ATA186) for incoming calls and to divert to voicemail after 20 seconds. However, when the 20 seconds is up, asterisk answers (I know this because I have another phone on the same line as asterisk) but the SIP phone keeps on ringing for a few cycles. How can this be? Can it be stopped? I'm running last week's cvs. Iain

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

A colleague called me through my * system via FWD using SJPhone and the quality was distinctly poor - a lot of hum and delay. Looking at the debug log the codec used was miscellaneously numbered 0, 4 and 8. I thought I'd disabled 4 (g.723) but it appears not. My sip.conf has this: general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to

Whilst in a call using the mec3 echo canceller today I had period of about 20 seconds of speech distortion. It's hard to describe but to me the call sounded as though we were having the conversation in a bathroom with some extra noise bursts and echo thrown in. I could carry on the call, with difficulty, and my correspondent didn't complain of any noise at all. After that 20 seconds

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

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 tried running the festival app today with little success. I have a working festival installation that does TTS to the linux sound output perfectly. With * it just produces a sort of hissing sound. The length of hissing is proportional to the length of text string that it is given to speak. Since I'm running on a PPC system I fear the dreaded endian problem is to blame and that

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

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

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