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Hi! When I try to load the ztdummy driver via "insmod ztdummy", I get the following errors: /lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_unregister /lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_transmit /lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_receive /lib/modules/2.4.26/misc/ztdummy.o: unresolved symbol zt_register I'm using zaptel-1.0.8 and

Hi! I have two Asterisk installations, one being a 1.09 bristuff installation and one 1.08 installation. In the 1.09 installation I have the "Dial" command available on the CLI, in the 1.08 installation I don't. My question is now: was that a new feature in 1.09 or is it a bristuff specific thing or is my 1.08 installation simply lacking features? Thanks, Christoph

Hi, I am very new to Asterisk. I wanted to know how to retrive the Voicemails. I could see some voicemails assosiated with some extensions. Any ideas?? --------------------------------- How much free photo storage do you get? Store your holiday snaps for FREE with Yahoo! Photos. Get Yahoo! Photos -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! I'm searching for a 1-800 number that simply plays music for a long time (>3mins) and no one picks up. I've bothered the AT&T lines so far when trying out my SIP->PSTN connection but then always someone answered :-) Anyone have a number? Christoph

hi all i suggest to create a goup in hotmail in order to discuss any problem on line in msn i think it's more practical than e-mail group Protek-on: CaraMail met en oeuvre un nouveau Concept de S?curit? Globale - www.caramail.com

Howdy, I''ve read on http://revu.tauware.de/revu1-incoming/xen-0604251805/xen-3.0.2/docs/misc/xen_config.html about the maxmem option? What it do? I don''t see any result... I ''ve thought that the memory will automatically scaled to maxmem... but xm top show only 128 mb reserver for the guest machine also during high loads. greetings Fabian

Howdy, 1.is it possible to install Windows 2000 in an guest machine? The setup loops forever on the "Windows 2000" is starting screen after copying the installation files to Harddisk.... I''ve also tried to press F5 during installation and choose Standard PC. ACPI,APIC is disabled... 2. Is it possible to install Windows NT 4? The setup hangs during installation... greetings

Hi all. I'm looking for an editor for sieve-scripts. I tried the Squirrelmail plugin "avelsieve", but it requires "timsieved", wich Dovecot doesn't use. I don't want my users to write sieve-scripts on the console and server-side-filtering is absolutely required. I know that there are some frontends available, but I can't figure out, which one really works

Hi! I would like to use the realtime extension of Asterisk and got the latest asterisk-addons from CVS. Upon compiling things, I got a couple of error messages from app_addon_mysql... is it me, or are the files in the CVS broken? Thanks, Christoph

Hi! I have a problem compiling the res_config_mysql.so after successfully compiling and installing asterisk with the bristuff package/patches. I get lots of compiler errors. When inserting a previously compiled res_config_mysql.so into the bristuff-patched Asterisk, I get an error on startup and Asterisk crashes. Anyone using res_config_mysql.so with the bristuff package and can help me? Thanks,

I'm very interested in the redirect feature of Asterisk. So far I haven't gotten it to work. My scenario is that there is a two party call going on where I want to send one of those parties somewhere else. In the wiki is only an example how to send both parties to a meetme room. Is the ExtraChannel parameter required? This is what I have: Action: Redirect Channel: SIP/8080-e2a7

Is there an option txgain for SIP in Asterisk? My users all complain that their other parties think that they are way too silent even though they all have their mic volume all the way up and also enabled the 'mic boost' option. This happens with all the clients that we're using and also with different model headsets, so my last hope is txgain for SIP. Thanks Christoph

Hi! When I want to monitor the OSS/dsp channel through the Asterisk management interface, I get a "permission denied" error: Action: Monitor Monitor: OSS/dsp File: 1124096949 Mix: 1 Response: Error Message: Permission denied My permissions for /var/spool/asterisk look like this: pound:~# ls -la /var/spool/asterisk/ total 40 drwxr-xr-x 10 asterisk asterisk 4096 Aug 9 10:19 .

Hi! I've read in the archives that there are problems concerning Nikotel calls being disconnected after two minutes. I had the same problem yesterday. Is there a fix? There was only a "giving up" statement after the last e-mail in the archive, I'm about to do that too. Here's my sip.conf entry for Nikotel (left out the register stuff 'cause it's working):

Hi! I have a strange problem. In an AGI I tell Asterisk to playback a number, for example 31. I then use the AGI SAY NUMBER command and I only hear "thirty" and then get: -- Playing 'digits/30' (language 'de') Sep 29 11:47:40 WARNING[3401]: file.c:475 ast_openstream: File does not exist in any format Sep 29 11:47:40 WARNING[3401]: file.c:787 ast_streamfile: Unable