similar to: newbiew extensions.conf question

Is anyone out there using app_dbodbc (http://www.bkw.org/~brian/app_dbodbc.c)? Any problems with it? I was able to get it all working, but it causes * to segfault every now and then. It does not appear to be related to any specific function (ODBCget,ODBCput,ODBCdel,ODBCdelltree). It is 100% repeatable. If I noload the module, everything works fine, but when its running, after calls to any of the

Could any of you provide me information on a good Polycom phone dealers to utilize. One who provides firmwares ..etc Thank you! Kenny ______________________________________________________ Click here to donate to the Hurricane Katrina relief effort. http://store.yahoo.com/redcross-donate3/

I can understand that it''s easier for rspec to set up a mock in advance of the controller call. But it makes it difficult to do something like: context "The HarkController, given Louie the logged-in user" do setup do post :login, :username => ''louie'', :password => ''atest'' end specify "should redirect Louie to the home

Hi all, I'm looking for some serious help. :) I couldn't find a better description for my problem... I think it is quite complex! Here's what I would like to achieve: A SIP caller dials into to my Asterisk 10. He will automatically listen to a specific MP3 stream. Other SIP callers dial also into my Asterisk. They all will automatically listen to the same MP3 stream. All

Does anyone else have DTMF issues with SIPPhone? When calling into my DID, and entering, say, 1002. Sometimes it will recognize it properly (rarely), other times it will receive something different. Such as, 1102 or 1000, etc. Has anyone else been having these issues? I'm only accepting ulaw and alaw, and my relevant sip.conf information follows: [sipphone] type=peer

I'm getting these messages every 7-10 seconds. -- Registered SIP '532' at x.x.x.x port 52956 expires 60 -- Registered SIP '532' at x.x.x.x port 56988 expires 60 -- Registered SIP '529' at x.x.x.x port 51444 expires 60 -- Registered SIP '529' at x.x.x.x port 64044 expires 60 -- Registered SIP '532' at x.x.x.x port 52956 expires 60 -- Registered SIP

Dear Developer, I'm having problem running ssh_keygen on my solaris 7 box. Can you please tell me as to why I'm getting this error as described below? I don't have that problem with solaris 8 that runs SMCossh 3.0.2p1 Thanks in advance. Louie # /usr/local/bin/ssh-keygen ld.so.1: /usr/local/bin/ssh-keygen: fatal: libcrypto.so.0.9.6: open failed: No such file or directory Killed #

Hi everyone, This is my first attempt at a PV guest install from scratch and I''m a tad confused. Have I answered one of the questions virsh-install asks wrongly? Does anyone know of a general guide for installing PV guests with virsh-install? (Most Centos docs seem to use the GUI based virt-manager and I only have command line access). Am I meant to make use of the files here:

https://bugzilla.samba.org/show_bug.cgi?id=3839 Summary: The '/' in the Softlink is missing on the remote Rsyncd server ! Product: rsync Version: 2.5.7 Platform: x86 OS/Version: Windows 98 Status: NEW Severity: normal Priority: P3 Component: core AssignedTo:

Hello User-List, is it possible to play a sound file directly to a caller channel? Like this AMI command Action: Originate Channel: SIP/20-00001d41 Application: Playback Data: /path/to/audio/file I get an Error Message. My intension is to play a sound file to a caller and the other callers don't hear this. Can someone help me ? Thanks a lot Bye Daniel

Hi all, How can I playback a file within an active call? I've tried with ChanSpy whisper mode like this (using AMI): Action: Originate Channel: Local/9999 at default Priority: 0 Variable: MSG=test Application: ChanSpy Data: SIP/1234-123 Async: 1 and in the dialplan: [default] exten => 9999,1,Answer() exten => 9999,n,Wait(2) exten => 9999,n,Playback(${MSG}) Where

Bradley Taylor shot me this *goldmine* of ugly ugly ugly dogs: http://www.sonoma-marinfair.org/uglydogvote.shtml I *have* to use one of these for the Mongrel BUGS Mascot. He''ll go on our bug list page and replace the little beetles on the left. Pick the dog you think best represents a lovable but defective pooch and reply to this with your +1. Let the voting begin! -- Zed A. Shaw

If any of you have been so bold as to install the Shorewall CA Certificate in your browser(s), the current certificate will expire on 11/13. There is a new 10-year certificate available for installation at: http://lists.shorewall.net/Shorewall_CA_html.html -Tom -- Tom Eastep \ Nothing is foolproof to a sufficiently talented fool Shoreline, \ http://shorewall.net Washington USA \

I am grouping my extensions by building like so: 1XX is Building 1 2XX is Building 2 7XX is Office [Office] extensions has the following includes 7xx Include => Local Include => International Include => Building1 Include => Building2 [Building1] has 1xx Include => Office Include => Building2 Include => Local I done't want building1 to access international, but does

Hello, Where I can find more info on how to nginx and mongrel boot when the machine reboots ? In os x, please. In Linux world I found some scripts but they don''t work and don''t go in the same place as OS X. thanks, r. -- Posted via http://www.ruby-forum.com/. --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the

You are right. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tarpo, Louie Sent: Thursday, August 11, 2005 5:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Newbie Question: Building anAsterisk systemtoreplace an old PBX but using existing phone You write out a

Dear all, Anyone of you knows how to play an audio file at the beginning of a call for both Caller and Callee? A(x) of Dial application only plays audio for callee. I don't want to use MeetMe because I want to use Monitor and MixMonitor. Thank you! ________________________________ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n

I have put together this call forwarding and dnd config: I'm sure it can be dome with macro's but I couldn't figure that out... anyone care to input. 74 Turns DND on my phone will not ring, drops caller to voicemail... 73 Turns DND off 72+ext forward your extension to another extension and voicemail is left at the forwarded extension. 71 turns off call forwarding. ; dnd Could

How does one do translation for calls that come in from other pbx's where the incoming caller ID is an internal extension number on their pbx? Eg. when I get a call from Free-World-Dial the CID shows up as "429102" which is essentially their internal extension number sans any routing prefix. To dial the number back I need to dial the extension with FWD's routing prefix

Hi all... I want to build my own domU and dom0 kernel. But when i do a "make world" I get something like "invalid python installation" and "can''t find /usr/lib/python2.3/config/Maikefile". I found the same problem in "xen-users" archive but the server is so slow that i can''t get the post. Only Google shows me that it''s there.