similar to: Pitching Asterisk

Is it possible to set asterisk up as a virtual pbx like in apache and virtual host? If so can someone point me to the right direction. I would also like to setup asterisk with some type of redundancy, I have searched the lists and googled but havent really found anything, I would be willing to put together a paper if I had the info and make it available, through asterisk doc project or

Is there a way to get * to use wav files instead of gsm files for the voicemail, agents, and queues applications? Gsm does not give all the quality we would like to have, and we use no low bit rate codecs.

Anyone have any insight as to why voip-info.org has been up and down all day, and more importantly unreliable for the last month? I assume the bandwidth is being donated or something, but surely someone would be willing to donate reliable bandwidth as the knowledge hosted on the site (which is also donated!) is worth way more than the bandwidth. There is no doubt it is the best

What would be the simplest way to retrieve information form a CNAM database that provides http based query responses? Does an application or script already exist that does this? Basically, I want to do a wget of a URL that contains the callerID number as a variable, and assign the returned text to another variable which can be used to set the caller ID name. Any suggestions?

Hello all I'm trying to get my asterisk config to listen to multiple ports. This is since some clients have port 5060 blocked by their ISP. Does anyone know how to do this in sip.conf or if it is even supported? Thanks!

Can anyone point me to a sample or information on using MeetMe like this? Conference room is set up with 2 PINs, one for the moderator and one for the participants. Participants get music until the moderator joins (to avoid wild, un-moderated tangents). Call is ended and all participants are kicked out when the moderator leaves (or the moderator can kick everyone out via phone keypad).

Can anyone point me to a reference or sample config for bypassing a nailed up (point to point) t1 between two PBXs with asterisk and a pair of t1 cards? Right now I have 2 Nortel norstars connected to each other via a leased line t1. I also have a solid 10mbps low latency microwave link between the 2 sites. My goal is to run an asterisk box at each end with a t1 card and Ethernet card to

Has anyone been able to get the IP501 to discover the FTP server IP address (via dhcp or dns) and download 100% of the config from a provisioning server? We are still having to touch each unit to enter the ftp server address and password, as well as set many of the options that will not take from the config file. Have a sample config file you are willing to share? What is required in

Has anyone had success with the TDM400 in production? I have multiple boxes where these cards lock up and the only thing that will fix them is to unload *, modprobe -r wctdm, modprobe wctdm, load asterisk. Does not matter if it is a FXS/FXO module. I know this topic has been discussed many times before, but my questions is not "is anyone else having this problem" since I know that

Has anyone had success with the Snom 190 displaying caller ID name and number on the Snom 190 on for an inbound call from *? Right now our Snom's only show the caller id name, not number. I know the number is transmitted from the Telco and received by * since the number shows on the incoming call event at the * console. We are not setting the caller id in the extensions.conf, simply passing

Looking for someone with a successful experience similar to this; I have a need to cross connect a 3COM NBX PBX PRI interface to asterisk, but over a long distance. We do not need any IP connectivity and the solution requires G.711u audio so there is no benefit to using IP. Has anyone here successfully cross connected any PBX PRI interface expecting NI2 PRI signaling B8ZS/ESF with an

I have added the following to a macro that is used for all extensions so a user can access voicemailmain by pressing * during the voicemail prompt ; check voicemail exten => a,1,voicemailmain(${macro_exten}) exten => a,2,hangup The behavior is a little weird, the * key is not recognized during the portion of the greeting where the extension number is being played back, after it is

Cant submit bugs - error 1303, invalid value for field when submitting a new issue. Bug info Failure on build with 1.2beta1 on fresh FC4 install ast_expr2f.c:1784: warning: no previous prototype for ?ast_yyget_column? ast_expr2f.c:1860: warning: no previous prototype for ?ast_yyset_column? ast_expr2f.c:1259: warning: ?yyunput? defined but not used gcc -g -o asterisk -Wl,-E io.o

Can anyone shed some light on "rules" that might make the task of parsing the country code and city codes from a dialed number in the CDRs? I know that there is almost never a case where a concatenated country and city code could overlap with another country code, but what about city codes and local numbers? Is it possible for a concatenated city code and local number to match another

Hi, I want to know how I can get a value from a table. Say, I have a table sip_buddies for storing sip user account information. There is a field called 'accountcode' that I want to get its value in the dial plan. As I find that there is no direct way to get the value from the table. Does anyone can tell me how can I get its value in the dial plan? Thanks!

It is amazing to me at this point that there is not an official Digium list of supported servers (including 1u models!). Clearly the number 1 issue with the Digium PRI cards is the server that they are used in. The new cards even go as far as listing server that DO NOT work on the Digium site! The wiki references are old and do not have any testing parameters. C'mon guys! Certify a

Anyone know how to override the 99 message limit in voicemail? (yeah, we have a public VM that gets that many a day). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050812/1bff12e4/attachment.htm

I am using realtime mysql for extensions, sip, and voicemail. Outbound call routing does not really perform well in realtime extensions due to the high number of rows in the database (300k), so I can not use it. It appears with my limited knowledge that the query method is not robust enough for large databases. Given the fact that I already have realtime and mysql configured, what are my options

Hi! A few months ago I needed some help for the following issue: .) a call comes in .) Person A takes the call and does an attended transfer to Person B .) Person A hangs up the phone without waiting for Person B taking the call .) the caller get lost at this point !! At this point the attended transfer should go into a blind transfer. The phone of Person B should still be ringing and the

Hoping someone can help me understand what is happening here; we start asterisk as a service at boot (actually, with heartbeat) on CentOS using the asterisk init script installed with "make config" upon reboot of the server (when the asterisk service is first started by heartbeat) we get color in the console when we connect to it using asterisk -r after the execution of