similar to: Call Transfer Features

Hi, Here i have been an matrix like this, *NAME AGE PALCE TRUE/FALSE* ABC 20 INDIA XYZ 30 FRANCE PQR 40 USA MNO 30 KENIYA DEF 25 AUSTRALIA GTY 34 CANADA BNH 38 JAPAN Here, *TRUE/FALSE *Column containing empty values. So my requirement what is, need to change all the TRUE/FALSE

Hi, Here i have been a matrix like this, *NAME AGE PALCE TRUE/FALSE* ABC 20 INDIA XYZ 30 FRANCE PQR 40 USA MNO 30 KENIYA DEF 25 AUSTRALIA Here,* TRUE/FALSE* Column containing empty values. So my requirement what is , need to change all the *TRUE/FALSE *column value into "*TRUE*" where *AGE =

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?

hi all, got a small prblem here Running Samba 2.2.8 with a couple of dell laptops that are used either standalone or connected to a network. Both users are storing lots of pictures on their C drive which are visible through My Documents/My Pictures. The problem is that when they are connected to the network, all these files are stored as part of their profile on the server. When it gets to

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!

Every few seconds I get the following message: == Parsing '/etc/asterisk/manager.conf': Found == Connect attempt from '127.0.0.1' unable to authenticate I'm trying to track down where it's coming from. I've used TCPDUMP & NGREP to monitor 127.0.0.1, no data's flowing. I've tried loading Asterisk with no modules, tried loading with a naked

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

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!

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

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

Michel Send me the same output for a dial string that only sends the *31* Is this an ISDN line? What type of card/signalling/switchtype are you using? It looks as if the PSTN switch accepts the *31* and then hangs up so you can make the NEXT call with the *31* feature enabled. If so I assume the *31* feature will be enabled for the next call on the ENTIRE SPAN if it is an ISDN trunk group. If

Just curious... Does anyone know if any companies using Asterisk on embedded hardware (out at the customer premisis), such as the Soekris Net4801, to provide VOIP service? Doug.

Anyone know of any equipment that I can use to connect a laptop running asterisk to a POTS line (RJ11) ? Regards, Dovid __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

First of all, Exchange is a complete solution. Dovecot is a imap/pop3 server ( a good one, sure... ) So replacing exchange means to find an integrated solution. about the questions: public folder was removed in exchange. IMHO they made the right choice :) There is a different thing, need different tools and different client app. You can create shared mailboxes ( i think it's a little

Hi, my / is located on /dev/hda6 and my "personal storage partition" - the palce where I store my important files - is located on /dev/hda9. I've deleted a couple of files in /dev/hda9/xyz/abc. Exactly I've deleted the whole directory "abc" with mc. Now hda9 is unmounted, of course. Can you give me a recipe about how to recover those files, please! Probably

I've read pretty much in the samba-howto-collection and in the manual for smb.conf and i now feel ready to compile and install the latest version of samba. But I have some question beacuse i had some problems before. My current version of samba is 2.2.3a-6 for Debian (apt-get install samba) and the reason I want to install the latest version is beacuse my friend did all the work for me the

This has been an interesting discussion for me (except for the sniping). The last post led me, out of curiosity, to this wiki entry: http://www.voip-info.org/wiki-Asterisk+TDMoE I was unaware of this feature, and it looks pretty good. I've been pondering replacing some T1's by leveraging IP capacity but of course have run up against the QoS issue. My idea was different... I

In an earlier thread Andrew Kohlsmith enlightened me on the use of qualify in sip.conf to deal with a peer that is down. Since then I have been searching for information on how the behavior of qualify can be tuned. The wiki is vague on this; " Syntax: qualify=xxx|no|yes where XXX is the number of milliseconds used. If yes the default timeout is used, 2 seconds. If you turn on