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Ongoing Project - sipXecs (was sipXpbx) PDF Print E-mail
Written by Administrator   
Updated March 14, 2008
 
Our Open Source project from 2005 forward was actually chosen in July 2005 after our internal tests with the Asterisk platform made us want to wait a while for a more mature administrative interface. We also were hoping to find a platform that was more standards centered as Asterisk used some proprietary protocols instead of industry standards. Our opinion at that time was that the Asterisk project had many advanced features, but that the management and configuration interface was too cludgy for a typical environment (too much room for error in its approach with manual editing of .conf files). This is just our opinion, and there is lots of work going on all the time to change that project. The best features in Asterisk are paid addons that really end up proprietizing the system, which is really what we are trying to get away from (not to mention they are expensive and leave you with dealing with a west coast company for support for the entire system).

With the sipXecs, we were able to come away after two weeks (basic testing, configuration and limited use) with the feeling that if someone wanted to implement (or develop) an open source Linux based SIP based application that was stable, reliable and based on industry standards, they should take a look around at SIPfoundry.

SIPfoundry has a lot of different projects, we deployed the sipXpbx project as a test of how well a number of their projects work together.

To us, this approach is distributed and we liked that aspect a lot. The PBX talks SIP. All of your devices external to the PBX talk SIP (hard or soft phones, gateways, ATA adapters), independently converting their input/output from speech to SIP or SIP to speech. The server can handle quite a load since its not really burdened with conversion tasks. GOOD JOB!

sipXpbx is part of a framework that lives at SIPfoundry. Check it out. The project as a PBX has changed its name to sipXecs, which includes a lot of well rounded features, like ACD functionality.





About SIPfoundry


SIPfoundry is an international open source software community dedicated to accelerating the adoption of SIP based VoIP solutions.

Founded in February 2004, the SIPfoundry developer and user communities have rapidly expanded to include participants from 63 countries and counting. SIP is hot, so whether you are a user or a developer get involved and consider SIPfoundry the place where you find or develop your SIP capabilities.


What we started with...

Within a few hours (from bare metal) we were able to install the OS (FC4, basic install,ISO CD #1 was all we used, we did add Webmin for some additional ease of use functions to gain access to the OS), sipXpbx (using the install script for FC4 from SIPfoundry) and a couple of software based telephones to make calls to and from each other. We now run FC6, Centos 6 and Suse, other distributions are available too. It does support Microsoft Exchange 2007 as the voice mail system or uses its own system, and can connect to LDAP to synchronize user data.

Later we implemented an AudioCodes MP104FXO gateway (a box that converts up to 4 POTS lines to SIP to make your link from the telephone company to the PBX itself) and had that working as well. We had problems getting this to work, and after posting a message to the sipx-users mailing list (must register to get these mailings, but well worth it!), we were rescued by Michael Picher of Central Maine Communications who helped us get our AudioCodes gateway to a point where we could figure out what else we needed to do. In our example, we posted this ZIP file with 46 screenshots that should help get you through a basic installation of this device. We've since returned the favor to invidual requests for help with this same sort of issue recently, since we believe that's the way this sort of thing should work.

You should have access to a DNS server to input the SRV records. If firewalling is desired, you should have iptables/patch-o-matic/conntrack experience handy as well as basic networking, linux and windows skills to make everything work in a normal environment. Basic telephony knowledge would be good in this environment too.


Caution needed by newbies...

A cautionary word... though we favor this approach we've outlined, there are many others available. For a test environment, we found this project to be superbly aimed at geeks who are in charge of corporate communications and have an aging or legacy type PBX that needs "overhaulin'" (ours didn't).

What we used in this test...

- HP DL380 server with 2.4GHz processor, 2GB RAM, 3 36.4GB HDD's (hardware RAID 5)
- Two Polycom 601 IP telephones
- One Grandstream Budgetone Model 100 Phone
- One SJLABS SJPHONE (softphone/WIN32)
- SJLABS SJPHONE for PocketPC (softphone/PPC)
- One Counterpath xten-lite (softphone/WIN32)
- One Sipxphone (softphone/WIN32)
- One AudioCodes MP104FXO Gateway

We used Windows XP PC's for the softphones (one PocketPC 2003 which was an iPaq 6315/cellphone/wifi/bluetooth model).

What we did outside the norm... we installed a few other packages into the Linux distribution to allow us to play with a STUN daemon.


What we REALLY liked:

1. Good how-to's on the Wiki (which was completely updated in March 2006).
2. Very stable, easy to add/delete users.
3. For a basic environment, it's really a no-brainer compared to other open source projects.
4. sipXconfig uses POSTGRES DB to store it's configuration and has an automated backup routine. The backup program does its job... we added (via webmin) to .tgz the backup folder from the pbx each day and transferring it to another server via SSH for archiving/restore purposes.
5. Pingtel (SIPFoundry Sponsor) is in the software business. This means they are not using their software code to sell their own proprietary hardware (unlike Digium/Asterisk, Nortel, Avaya, Cisco, NEC, 3COM).

What we disliked:


1. The documentation on how to make the AudioCodes gateway device configurable from the sipXconfig server was really not all there, so we used a manual configuration. If we get enough information on how to make it "play nicely" in this manner, we'll submit to SIPfoundry. In 43.8 (BETA in April '07), the plug and play support for Patton Gateways will be added. We already have one to test with!
2. There are no inexpensive T1 gateways/cards. This is where the Asterisk project has their own "proprietary" (albeit inexpensive PCI card, only 595.00) hardware that fills a huge void. Expect T1 interfaces of the SIP variety to cost in the 3,000.00 range (these can be cards or standalone boxes).
3. Limited manufacturer "config push" support for gateway devices. 

What we REALLY wished it had:

1. At least a POP3 (IMAP4 would be preferred) connector to it's voice mail message store in lieu of an email "forward", which would get you an equivalent "unified messaging" approach instaed of calling a forwarded email "unified".  We think a slightly modified Postfix mail server on the same system would work very well since it is Postgres friendly, though it would have to use the IMDB for autenticating of users and read the XML file for the mailboxes for the users. If this can be done as an IMAP server, it would be very much "integrated" by the email client as another account (listen to the voice mail on the PC or the phone, delete the message from either and watch it disappear in both devices meaning the inbox and the message waiting indicator on the handset).
2. Time based greetings for users. It does not appear that users actually have their own attendant configurable from the system.In version 3.4, an Auto Attendant Dialing Plan was implemented in sipXpbx. which appears to be very robust and includes a holiday schedule. There is no time based user auto attendant (greeting schedule) yet. The user can dial in (or from a web browser) and change the greeting by re-recording it or to use a different recorded greeting as the "active greeting.
3. Better ACD configuration and options, it's very basic. What it does have seems to work well in our test environment. There is purported to be a more robust ACD in the commercial version offered by Pingtel.

As for FAX integration, we really don't embrace the notion of integrating that into a PBX environment. Your options for IP telephony really open up when you leave FAX out of the equation, at least that's our opinion right now. That's different than hooking up a FAX to a FXS or ATA device to use it, which is fine, but what we're talking about is having the PBX receive and render/rasterize the fax seems a bit difficult to do in an open and standard way just yet. Check out our diskless fax choice in another review.

What's currently being done by us with sipXpbx?

We use it with our RT (using version 3.10, ticket queues updates workers via RSS feeds from anywhere) helpdesk to automatically create helpdesk tickets and add the voice mail as an attachment. It can notify us in the field (we use a GPRS enabled cellphone, download the voice mail and listen to it on Pocket PC 2003 WITHOUT converting or docking. Check out TCPMP (The Core Pocket Media Player) previously called "Beta Player". We've always been confused why you have to dock a PocketPC and let Windows Media Player at the desktop convert a .WAV file encoded as PCM or uLAW to bring it to the handheld. No wonder Microsoft stopped calling it Pocket PC. Why DO they now call it Windows Mobile (not!)? It works natively with playing from our HP iPaq 510 voice messengers or Blackberry 8830's.

UPDATE - Since sipXpbx version 3.6, the wave file format works fine with the embedded Windows Media player on Windows Mobile Phones.

We use SIP TAPI to allow click2dial applications (Act!, Goldmine, GroupWise, Outlook) to dial through the sipXpbx (no modem required), on Windows PC's. This is not a full TAPI driver in that it does not give you screen pops of incoming calls or lookup contact records and bring them up when someone calls in. It is a simple outbound dialer and it seems to work OK. 

What other ways can sipXpbx be enhanced?

Can you use WEBDAV to place a folder on the desktop for easier access directly into the user's voicemail store? Should you? Right now, with issues in the Webdav environment, we feel it's best to leave this alone for right now. Once Webdav is a little further along, it should be considered.

We tried to integrate analog DID from our local TELCO (worked great with our old phone system). Unfortunately, Embarq executed someone else's work order on our lines took our service down by accident.  They could not get the service back they way it was originally ordered , and as our phone service was out for several days due to their error, we felt it prudent to abandon our DID numbers of several years and revert to POTS lines. The TELCO could actually deliuver these without issue, though we had to place the order a second time as they failed to activate some special options. Since then we have partnered with Ingate and Bandtel, to bring ITSP services through our firewall, and are able to deliver all digitial DID service at a faction of analog or traditional T1/PRI service. 

As a matter of fact, we actually use our preferred analog gateway models from Patton electronics (Smartnode) to connect directly to our Multitech FaxFinder for inbound DID calls. We set aside an FXO port and add some specific rules, and all our fax calls go to the MultiTech with DID-to-email routing. We also connect a Cisco ATA adapter and dial out via the PBX from the Multitech FaxFinder while in the office or on the road via our VPN. Works like a diskless champ!


Instead of having many different dialing plans at our analog gateways to route local and long distance calls, we opted for multiple analog gateways initailly. Initially, in our environment, we used one gateway for local lines, then another to plug in an ATA adapter from Lingo (any VOIP provider will do, Vonage, Lingo, Packet8, etc.). We then told our sipx system to use one gateway for all local outbound, toll-free and emergency calls. We told sipx to use another gateway (where the Lingo adapter was plugged in) to handle outbound Long Distance calls to avoid toll charges, way cool. Until we immesed ourselves in the Ingate SIParator product, it worked OK. Once we started using the Ingate and got true digital service, we never looked back.

Testing of the new MultiVoipSS (MultiTech's VOIP Gateway with a "survivability" feature) began in May 2006. The idea behind this to enable the "main" or "branch" site to have a fail-safe inbound/outbound call system in the even communication to the main SIP server is unavailable. We are proposing to set up a branch site and sever the Internet connection making the "gateway" go into a fallback mode. Remote sites using sipXpbx could use a single gateway, all using a single PBX for voice mail, provisioning, etc.  While we've found thegateway useable, it was unstable and clearly needs better factory support. We've settled on Patton Smartnodes, they are inexpensive, dependable and HIGHLY configurable.

This page is always a work in progress. So check back!