search for: gdh

...Budgetdedicated APT repo on Ubuntu dapper... 0.9.17 works a treat, but 0.9.18 and 0.9.19 immediately call the debugger on /any/ app, even winecfg + /usr/bin/notepad.. I've tried deleting my .wine dir, and always ensure there is no wineserver hanging around, but always to the same end effect: gdh@plip:~$ rm -r .wine gdh@plip:~$ notepad wine: creating configuration directory '/home/gdh/.wine'... wine: Unhandled page fault on write access to 0x003ffd80 at address 0x60a7dc0e (thread 0009), starting debugger... wine: Unhandled page fault on write access to 0x003ffd80 at address 0x7e798c...

...set to No). In this way, I can preserve the state of agents between Asterisk restarts. However, I notice in 1.4 beta3, this application has gone. Can anyone suggest what would be the best alternative? I have thought of System(psql -h xxx -U xxxx -P xxx .......) but that's just horrendous :) gdh

...cube's UI to generate .dovecot.sieve files. We use winbind + LDAP lookups to do some exotic mail rewriting... ultimately user.name at domain.com gets rewritten to 'ar-cloh at localhost', and in /home/AD/ar-cloh/.dovecot.sieve there is only this: require ["vacation"]; # rule:[gdh_test] if anyof (header :contains "Subject" "test") { vacation "test away"; } When I send an email to the user, I see this in the mail.log Oct 19 15:39:43 ar-smtp-1 dovecot: deliver(ar-cloh): sieve: msgid=<1255963176.14420.31.camel at gdh-work>: discardi...

If extension 101 calls 102 and user 102 hits # and then 103, the caller ID of 103's phone says 102. I've been looking for a way to have 103's Caller ID show the person that is being transferred not the person transferring. So if my receptionist answers the phone and transfers it to one of my techs, I want my techs phone to display the caller ID of the person who called the

I have been impressed with the quality and meagre bandwidth of the G729 codec from Digium. I am in a testing phase of our roll out, we are using 5 Asterisk PBXs in various countries to provide connectivity for our employees, owners and family. As we are testing, and our setup is somewhat complex due to the peculiarities of our connectivity, there has had to be a lot of changes to servers, cards to

...Tuesday 31 May 2005 14:41, Giordano Grandis wrote: > Hi Gavin, > But...how atxfer work ? Ehm, just the way I explained yesterday :) Just make sure you include the 't' option to the Dial application, in the same way you need for the old-style '#' blind-transfer to function. gdh _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

On Wed, Aug 03, 2005 at 11:28:19AM -0500, asterisk-users-request@lists.digium.com wrote: > 10. Inter-Tel AXXESS failure: HDLC Bad FCS (8) on Primary > D-channel of span 1 (Gavin Hamill) > Date: Wed, 3 Aug 2005 15:32:48 +0100 > From: Gavin Hamill <gdh@laterooms.com> > Subject: [Asterisk-Users] Inter-Tel AXXESS failure: HDLC Bad FCS (8) > on Primary D-channel of span 1 > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> > Message-ID: <200508031532.487...

Hi, I've used Agents + Queues before with success, but I can't figure out why this trivial setup is not functioning... stage*CLI> show agents 1306 (gdh) available at '1306@internal' (musiconhold is 'default') 1 agents configured [1 online , 0 offline] and the internal context is simply: [internal] exten => _13XX,1,Dial(SIP/${EXTEN},,h) Now, taking this line... exten => 123454,1,Dial(SIP/1306) (Legacy PBX On Zaptel interfa...

...ata/profiles writable = yes create mask = 0600 directory mask = 0700 Any ideas, you lovely people? :) I also have the Win2000 printer status 'Access denied, unable to connect' problem, but I'm still tinkering with that before I register an offical cry for help ;) Cheers! gdh

I known. I'm using the 1.44 firmware version relesed on 26 may. I worked for italian IVR an HTTP pgaes. So i can only update asterisk with CVS and try atxfer. Thanks for all -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Gavin Hamill Inviato: luned? 30 maggio 2005 18.40 A:

Hello, I'm trying to use the nsswitch module. Everything compiled fine and I was happy to see my NT-domain-users with getent passwd. BUT getent passwd MYDOM+myuser doesn't work. ls -l returns only numeric uid's for domain members. After I tried a ls -l , getent passwd returns only the local users for some seconds. If I try it again 30 sec later I get the complete list again. I

If you log in you will be a Windows 2000 standard user. Some applications need to run as poweruser. How can I configure samba to log in certain users as powerusers? Volker

Hullo. Having found http://bugs.digium.com/bug_view_page.php?bug_id=0000875 I grabbed the original 0.0.1 and Dan's patch, and whilst it didn't apply, I was able to patch the zaprtc.c manually - the Makefile has changed a lot, and I wasn't able to understand the changes. (this is all on a machine that's never had any * on it before, and has a 2.6.5 kernel with a matching

...ator, etc. However, I cannot get MX2004 [1] installed. I'm running Wine 20050211. If I run the setup in Windows='win98' mode, it launches perfectly and politely tells me that Win98 is not a supported operating system. So, I change to Windows='winxp' and get this ugly error: gdh@plip:/tmp/Disk1$ wine Setup.exe wine: Unhandled exception (thread 000d), starting debugger... err:dbghelp:pe_load_dbg_file -Unable to peruse .DBG file ole32.dbg ("E:\\Disk1") Does anyone have ideas how I could solve this problem? Cheers, Gavin. (and sorry for the half-post - hit ctrl-en...

Hello, I have a little trouble when entering into a domU console (xm console mydomU) : i can''t use the CTRL-C sequence to stop a script/command (like a continuous ping for example) Is there any parameter / tip for that ? Arnaud _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, As I understand it, IAX2 trunking requires zaptel timing. Zaptel timing is provided by Zaptel cards, ztdummy or ztrtc. ztdummy requires usb-uhci and ztrtc can't run on smp systems. So if you only have smp systems with ohci and no zaptel cards (because it's a sip/iax2 gateway) then you're screwed? - -- Regards, Tais M. Hansen

Hi all, I installed all needed RPMs by GnoPhone to be installed without problems but when attempting to install GnoPhone itself I get this message: # rpm -Uvh gnophone-0.2.4-1.i386.rpm error: Failed dependencies: mozilla >= 0.9.2 is needed by gnophone-0.2.4-1 libgtkembedmoz.so is needed by gnophone-0.2.4-1 libgtksuperwin.so is needed by gnophone-0.2.4-1 I'm using

Hullo :) Please be gentle with me, I don't have a working * install, and am just looking for background information. I'm always impressed by companies who implement a queue like "You are now number N in the queue. There are currently M agents answering calls, and your call should be answered in approx. O minutes" I've seen on

Hi Can anyone provide me with a Manchester (0161) UK DID number, preferably IAX2 but SIP is ok too, that I can use for my incoming calls? Call volume will be low. The critical thing is that DTMF must be correctly passed 100% of the time, unlike Sipgate, my current (free) provider, whose DTMF detection/passing is not at all reliable, making it useless for a virtual receptionist scenario. I

I am about to buy AT-320EE phones for use together with asterisk. Has anyone used these phones ? Are there any known problems using them using Asterisk ?