search for: lithiumblue

Displaying 5 results from an estimated 5 matches for "lithiumblue".

2007 Nov 08
2
asterisk and installing chan_h323.so rpm
Hello, When I tried to install chan_h323-1.0.1-module.i386 RPM i got these errors. Failed dependencies: libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 But i found the same files in /usr/lib/libh323_linux_x86_r.so.1 /usr/lib/libpt_linux_x86_r.so.1 What to do for asterisk to detect the same
2008 Feb 05
6
External MWI question for Asterisk
Hey there. I've been working on a project to integrate Asterisk with Exchange Unified Messaging via sipX using large parts borrowed from: http://blog.lithiumblue.com/2007/04/accessing-exchange-2007-unified_29.html ... and everything works surprisingly well. The one problem I have is MWI, or a lack thereof. Exchange 2007 doesn't support MWI of any kind (!), so I've been looking into using a product from Geomant to fill that gap. They have a package...
2008 Nov 22
1
IMAP voicemail with Exchange (was: A way to run extenrnotify when IMAP events take place...)
Hi Jeff - > I have IMAP voicemail working with Exchange 2003 using a single username and > password for multiple mailboxes. Sorry to hijack this thread (at least I changed the Subject), but this really caught my eye. I was under the impression that Exchange's IMAP doesn't have the master user feature and therefore can't do single username authentication for multiple mailboxes.
2007 Sep 21
3
Asterisk and MS Exchange 2007
Hi, Here you can find a list of MS Exchange 2007 compliant systems: http://www.microsoft.com/technet/prodtechnol/exchange/telephony-advisor.mspx You cannot see mention of Asterisk ;-)) Has anyone tried to "integrate" Asterisk and Exchange 2007 ? A prospective customer is using MS Exchange 2007 and is asking for such integration. Passion aside, what do you think of such integration ?
2007 Nov 11
1
RTP traffic not being forwarded
Hi, I currently have an issue where asterisk is not forwarding the RTP traffic between 2 endpoints. The SIP session gets set up correctly, and both parties get connected. The RTP audio is being sent by both endpoints to the correct ports on the Asterisk server as per the session description in the SIP conversation. However, Asterisk is not forwarding either endpoint's RTP traffic to the