similar to: Suppressing CallerID in .call files

Hi All, I am trying to get Asterisk up and running on my new Mandrake 9.1 install. I've installed Linux in the "standard" mandrake security mode, and "su" to do my attempts at install. I managed to obtain the source from CVS, and have been able to compile Zaptel. I then ran insmod zaptel, and also make config. I think I have compiled and loaded Zaptel successfully as

Hi! A customer of mine wants to connect an asterisk system with 240 to 480 lines to a PSTN switch. To save the costs for E1 cards and the corresponding E1 mainlines he wants to connect the system to the switch by a SIP trunk. Phones will be connected to the server through the same SIP trunk as this will be some kind of a "hosted pbx". Given he finds a provider wich has this much SIP

I've tried the precompiled version of gastman, but it does'n work properly under windows, so I would like to try to compile it under windows, so maybe it begins to work. I've tried to compile gastman with mingw, but the db3.1 libraries request the cygwin include files and libraries. Then I've copied the cygwin's include files and libraries to mingw's directory, but the make

Hi all, Are there any helpers or so for listview routines that suppres the three above mentioned columns? I''m currently exploring something like the code below. But it feels like I''m doing Rails'' work. # just for test purposes <div id="debug"> <% for col in Product.columns.map %> <%= col.name %> - <%= col.type %><br/> <%

I'm trying to connect my Asterisk server via sip using my vonage soft phone account. Has any anyone successfully got to work? I get error from asterisk saying: == Parsing '/etc/asterisk/sip.conf': == Parsing '/etc/asterisk/sip.conf': Found Aug 24 11:01:11 WARNING[1125329600]: acl.c:146 ast_get_ip: Unable to lookup '216.115.25.199:5061' when trying to register with

Hi! Does anyone of you have an idea how to detect an answering machine on a dialout call? I am working an a voicemail system wich calls the subscriber but I don't want to fill their answering machine. Maybe I could detect somehow if there is incoming voice when playing the message. usually real persons don't talk when they listen but answeringmachines do. ;-) Thanks in advance,

Can anyone tell me the features of the Cisco 7905 with SIP? I mean things like number of lines, speakerphone, transfer buttons, etc. I've seen the Cisco material, but all it told me was how nifty it is and how wonderful the XML interface will be ;) Thanks, --Ernest

Full_Name: Shane Neph Version: 2.2.1 OS: mac os x Submission from: (NULL) (71.113.43.247) While I found the names of the package authors and maintainer, I was unsuccessful in finding any contact information. The preliminary documentation for heatmap.2 is inconsistent in at least a couple of places when discussing the suppression of one or more dendrograms (and column/row ordering in general).

Very friggen cool, that you very much for the information it looks like it will do the job nicely. What did you use in your extensions list to activate the relay? James > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jay Milk > Sent: Sunday, 20 February 2005 6:24 PM > To: 'Asterisk

I download app_valetparking.c from http://www.loligo.com/asterisk/misc/apps/app_valetparking.c and followed the directions on http://www.loligo.com/asterisk/misc/apps/app_valetparking.README I am using asterisk-1.0.0 any suggestions [root@localhost asterisk]# astxs -install apps/app_valetparking.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat ions -g -Iinclude

The chan_sccp page at http://www.voip-info.org/tiki-index.php?page=chan_sccp2 has been updated. See the bottom of the page. Thanks. Comments welcome. -- respectfully, Joseph =============== ---------------------= ********** =

Hi people, what is the current status of send/receive fax on asterisk extensions, i dont want to receive the fax and send an email or viceversa, i want to connect a standard fax machine to a Linksys' ATA (FXS RJ11 port) . Webdoc?, pointers? Thanks

Hello all, I noticed that Cisco claims that you can do station to station intercom with the 7940/7960 phones, and the Cisco Call Manager. Does anyone have an example SIP header that shows this in action? Or is it something else that triggers the intercom? I would like to add this in to *. Thanks, Mike

Greetings, I am running samba-3.0.10-1.4E.9 installed from rpm on CentOS 4.4. I have it configured as a PDC. It is using the /passdb backend = tdbsam/ backend. I am using /pdbedit/ to make some configuration changes to user passwords. I would like to expire a users password, so that they are required to change it the next time they log in. From all that I have read in on-line resources

Hi, I was able to implement T122p one port PRI and was able to call out, but I am planning to use TE412p (includes echo cancellation) 4 port digital card (PRI), I wanted to know can asterisk support 3 four port PRI cards (12 PRI connections) with proper hardware like dual core quadcore processor and 8gb RAM in one server? Also I was planning to implement using 64 bit architecture with Asterisk:

Hi Is it possible (or recommended) to run both Asterisk and say SER on the same physical machine? How about port conflicts? Maybe the easiest way is to change the default SIP port on Asterisk? But how will that work if I register some SIP accounts directly from asterisk (like my SIP provider) but then wanna dial outbound pure SIP calls via my SER... Has anyone got a functional system like this

Hi,How to accepts the call and plays a voice message on the line without billing the caller ? This may be necessary for IVR applications that want to explain features of the service offered . Regards _________________________________________________________________ MSN 8 with e-mail virus protection service: 2 months FREE* http://join.msn.com/?page=features/virus

Dear Friends, We have implemented "Asterisk" in our organization. There are 150 members in our organization. At present all are using softphones. Now, I want to buy hardphones for our staff. Can anybody suggest me that what is the best hardphone for Asterisk with low-cost? Thank you. Regards, Chandra. --------------------------------- Ring'em or ping'em. Make PC-to-phone

Hi! Just ashort question - obviously I am too stupid too find the answer on the net. :-) I want to upgrade a running Asterisk 1.4.2 to 1.4.6 - what will I have to do? Just install it over the existing version? Do I need to backup the configuration? Will I need to reconfigure the source or will the new version "import" my old settings? Will I need to update Zaptel and Libpri too?

Hi all, I need your lights :) There are many hardware provider for E1 cards on the market, what's your exeperience with E1 and what's the preferred provider for Asterisk out of Digium? Olivier