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[ongoing] ongoing · A Sip of the Future: it’s a black box that uses, one hears,all sorts of clever tricks to skate around the realities of the Internet As ItIs to get the job done. I’d regularly heard people who knew something of theterritory asserting that this was evidence that SIP was a theoretical nicetydreamed up by IETF hippies but it wasn’t going anywhere.

[Ubuntu Linux] What is SIP | a community for beginners and experts: In order to provide telephony services there is a need for a number of different standards and protocols to come together - specifically to ensure transport (RTP), to authenticate users (RADIUS, DIAMETER), to provide directories (LDAP), to be able to guarantee voice quality (RSVP, YESSIR) and to inter-work with today's telephone network. Here we will only cover SIP.

[VoIP Forums - VoIP Call Termination, Services, Software and Hardware] VIPPIE Tunnel Mobile Dialer @ Amazing Prices by S4VOIP.com - VoIP ...: For higher traffic recommended is cluster configuration with two or three servers with voipswitch software running on seperate servers and connected to one, shared SQL database installed on a dedicated server. Web services running on a backup SQL server set in replication mode so that the load from web requests (for example browsing CDRs, statistics) does not affect the performance of the main SQL server.

[trixbox - The Open Platform For Business Telephony] sip extension problem | trixbox: I just installed Trixbox 2.6.1, my zaptel extensions are working fine when I called outbound. But when I use SIP extension to call outbound it connect to out bound and when someone receive my call then he/she never hear me while I can hear him/her.

[DownArchive - Your Future Downloads] Building Telephony Systems with OpenSER » DownArchive - Your ...: OpenSER is a flexible, free open-source VoIP server based on the Session Initiation Protocol (SIP), an application-layer control (or signaling) protocol for creating, modifying, and terminating sessions with one or more participants, including internet telephone calls, multimedia distribution, and multimedia conferences.

[trixbox - The Open Platform For Business Telephony] Trouble with Cisco 7940 Converting to SIP | trixbox: Trouble with Cisco 7940 Converting to SIP. rfscholl. Posts: 2. Member Since: 2009-09-21. Submitted by rfscholl on Mon, 10/05/2009 - 8:30am. I am new at this, so I followed the instructions found here: ... but working great!! 2 x Cisco 7940 (Bought on EBay. Could have got it for less if I had the patience...) Callcentric "Pay per call" and "DID Residential service" for In/Out calls. Login or register to post comments. nycmaster. Posts: 13. Member Since: 2009-09-30 ...

[JimmyRay's blog] Overlooked VOIP Security Features | NetworkWorld.com Community: Taking human behavior into account here, I am going to assume (dangerous, I know) that many VOIP networks are config’ed by data folks interested in Voice and not the other way around. Some of the old school phracky-phrack stuff could come in handy here.

[Dalmietron Solutions] Dalmietron Solutions: SIP technology for VoIP: In order to provide telephony services there is a need for a number of different standards and protocols to come together - specifically to ensure transport (RTP), to authenticate users (RADIUS, DIAMETER), to provide directories (LDAP), to be able to guarantee voice quality (RSVP, YESSIR) and to inter-work with today's telephone network. Here we will only cover SIP.

[VoIP User - IP Telephony Resource] Two Really Interesting Projects in Telecoms: The second thing I came across this week is a PHP based SIP stack created by Level 7 Systems Being PHP based this could really open up telephony applications for web developers. It's open-source and available from their Google Code repository here.

[Asterisk News] OpenSER Kamailio SIP Masterclass: Hello, The next edition of Kamailio SIP Masterclass takes place in Berlin, Germany, November 9-13, 2009. The training is focused on teaching Kamailio along 5 full days, up to advanced level, touching everything needed to build large and secure VoIP networks, integration with Asterisk media server as well as NGN-class of services for telephony and IP unified communication, such as instant messaging, presence, integration with social networking.

[mediacenterhouse.com] mediacenterhouse.com - a blog of custom installed Media Center PCs ...: Setting up the SIP account though requires quite a few more setttings but rather than do this through the small screen on one of the handsets (small but at least colourful and clear), it can be done through the base units default IP address in a web browser (192.168.0.2 for mine until I gave it a static address; important if you go down the route of using QoS settings on you router (Quality of Service - so calls are treated with the highest priority by the router to avoid data drop-outs during high bandwidth usage)).

[Jordy Blog] Custom Communication Apps | Jordy Blog: Some of the recent FreeSWITCH customization projects we’ve built for our clients include a custom call center that can handle up to 100 concurrent agents on commodity hardware, and a distributed SIP load tester that’s capable of pushing thousands of concurrent SIP calls (suitable for stress testing extremely large telephony infrastructures).

[Michigan Telephone, VoIP and Broadband blog] Review of FreePBX 2.5 Powerful Telephony Solutions by Alex Robar ...: One other point I should make ”” as the title of the book implies, it deals with a particular version of FreePBX, namely version 2.5.  Of course, as so often happens with a book about software, the ink is barely dry on the paper when a new version comes out.  FreePBX 2.6 has already been offered as a release candidate, and beta versions of FreePBX 3.0 are being made available.  From a user’s standpoint, version 2.6 will be nearly identical to 2.5 – there may be a few added options and such, but for the most part they are not things that you would need to worry about, or that would detract from the accuracy of this book.  However, FreePBX 3.0 will be a major rewrite, but it’s only available in an early beta version, and unless you are an experimenter that wants to be on the bleeding edge, you don’t want it yet.  Whenever you do move to FreePBX version 3.0 ”” and I’d be very surprised if a full release version is much closer than a year away ”” much of what you’ve learned about FreePBX 2.5 and subsequent versions will still be applicable (and also, I suspect that people will be using FreePBX 2.x versions for quite some time to come).

Reflected tags on Technorati: Blog, Sip Telephony, VoIPBlogging