Wednesday, December 31, 2008

Vyatta Rules to Block China / Pacific Rim IP's

Created a set of Vyatta rules to block all traffic from Pacific Rim nations. I usually do this as a matter of course with firewalls with customers. To get the addresses I combed through IANA's database.

The first set should be the first set of rules inbound on your outside interface.

set firewall name INBOUND rule 5 action drop
set firewall name INBOUND rule 5 source address 202.0.0.0/7
set firewall name INBOUND rule 5 log enable
set firewall name INBOUND rule 6 action drop
set firewall name INBOUND rule 6 source address 210.0.0.0/8
set firewall name INBOUND rule 6 log enable
set firewall name INBOUND rule 7 action drop
set firewall name INBOUND rule 7 source address 218.0.0.0/7
set firewall name INBOUND rule 7 log enable
set firewall name INBOUND rule 8 action drop
set firewall name INBOUND rule 8 source address 220.0.0.0/7
set firewall name INBOUND rule 8 log enable
set firewall name INBOUND rule 9 action drop
set firewall name INBOUND rule 9 source address 222.0.0.0/8
set firewall name INBOUND rule 9 log enable
set firewall name INBOUND rule 10 action drop
set firewall name INBOUND rule 10 source address 59.0.0.0/8
set firewall name INBOUND rule 10 log enable
set firewall name INBOUND rule 11 action drop
set firewall name INBOUND rule 11 source address 60.0.0.0/7
set firewall name INBOUND rule 11 log enable
set firewall name INBOUND rule 12 action drop
set firewall name INBOUND rule 12 source address 114.0.0.0/7
set firewall name INBOUND rule 12 log enable
set firewall name INBOUND rule 13 action drop
set firewall name INBOUND rule 13 source address 116.0.0.0/6
set firewall name INBOUND rule 13 log enable
set firewall name INBOUND rule 14 action drop
set firewall name INBOUND rule 14 source address 120.0.0.0/6
set firewall name INBOUND rule 14 log enable
set firewall name INBOUND rule 15 action drop
set firewall name INBOUND rule 15 source address 124.0.0.0/7
set firewall name INBOUND rule 15 log enable
set firewall name INBOUND rule 16 action drop
set firewall name INBOUND rule 16 source address 126.0.0.0/8
set firewall name INBOUND rule 16 log enable


This second set of rules should be set for inbound on your inside interface.

set firewall name OUTBOUND rule 5 action drop
set firewall name OUTBOUND rule 5 destination address 202.0.0.0/7
set firewall name OUTBOUND rule 5 log enable
set firewall name OUTBOUND rule 6 action drop
set firewall name OUTBOUND rule 6 destination address 210.0.0.0/8
set firewall name OUTBOUND rule 6 log enable
set firewall name OUTBOUND rule 7 action drop
set firewall name OUTBOUND rule 7 destination address 218.0.0.0/7
set firewall name OUTBOUND rule 7 log enable
set firewall name OUTBOUND rule 8 action drop
set firewall name OUTBOUND rule 8 destination address 220.0.0.0/7
set firewall name OUTBOUND rule 8 log enable
set firewall name OUTBOUND rule 9 action drop
set firewall name OUTBOUND rule 9 destination address 222.0.0.0/8
set firewall name OUTBOUND rule 9 log enable
set firewall name OUTBOUND rule 10 action drop
set firewall name OUTBOUND rule 10 destination address 59.0.0.0/8
set firewall name OUTBOUND rule 10 log enable
set firewall name OUTBOUND rule 11 action drop
set firewall name OUTBOUND rule 11 destination address 60.0.0.0/7
set firewall name OUTBOUND rule 11 log enable
set firewall name OUTBOUND rule 12 action drop
set firewall name OUTBOUND rule 12 destination address 114.0.0.0/7
set firewall name OUTBOUND rule 12 log enable
set firewall name OUTBOUND rule 13 action drop
set firewall name OUTBOUND rule 13 destination address 116.0.0.0/6
set firewall name OUTBOUND rule 13 log enable
set firewall name OUTBOUND rule 14 action drop
set firewall name OUTBOUND rule 14 destination address 120.0.0.0/6
set firewall name OUTBOUND rule 14 log enable
set firewall name OUTBOUND rule 15 action drop
set firewall name OUTBOUND rule 15 destination address 124.0.0.0/7
set firewall name OUTBOUND rule 15 log enable
set firewall name OUTBOUND rule 16 action drop
set firewall name OUTBOUND rule 16 destination address 126.0.0.0/8
set firewall name OUTBOUND rule 16 log enable

If you are using any local services on the firewall itself, consider applying the rules to the local section of the outside interface.

Vyatta with OpenSBC Install

Finally got pointed in the right direction with Vyatta and OpenSBC by Joegen (the main developer over at OpenSipStack.org). Here are my install notes for OpenSBC on Vyatta.

Install OpenSBC on Vyatta Firewall


Install Vyatta:


Boot from Vyatta LiveCD ISO. Press ENTER.

At login, username 'root', password 'vyatta'.

Enter 'install-system'

Configure as needed... I'll share mine when I get this all working...


Get items required for building OpenSBC (kitchen sink approach):


Login to Firewall as user 'vyatta'

cd /etc/apt

su

Password: (enter root password)

nano -w sources.list

Add line: "deb ftp://ftp.us.debian.org/debian/ lenny main contrib non-free"

Ctrl-X and Y to overwrite

apt-get update

apt-get install -y mc autoconf automake cvs flex expat libexpat1-dev libtool build-essential libxml2 libxml2-dev libtiff4 libtiff4-dev php5 php5-cli php5-mysql php5 php5-cli php5-mysql php5-gd mysql-server libmysqlclient15-dev php-pear php-db curl sox apache2 libssl-dev libncurses5-dev bison libaudiofile-dev subversion libnewt-dev libcurl3-dev libnet-ssleay-perl openssl ssl-cert libauthen-pam-perl libio-pty-perl libmd5-perl libpg-perl libdbd-pg-perl php5-pgsql sqlite3 libsqlite3-dev openssl ssl-cert libapache2-mod-php5 php5-cli php5-common phpMyAdmin php5-mcrypt mcrypt phppgadmin apache2 libmcrypt-dev


Get OpenSipStack and OpenSBC from CVS:

cd /usr/src

cvs -d:pserver:anonymous@opensipstack.cvs.sourceforge.net:/cvsroot/opensipstack login

cvs -z3 -d:pserver:anonymous@opensipstack.cvs.sourceforge.net:/cvsroot/opensipstack co -P opensipstack

cvs -z3 -d:pserver:anonymous@opensipstack.cvs.sourceforge.net:/cvsroot/opensipstack co -P opensbc


Compile / Make OpenSipStack and OpebSBC:

cd /usr/src/opensipstack

chmod +x ./configure

./configure

make bothnoshared

cd ../opensbc

chmod +x ./configure

./configure

make bothnoshared

make distrib

Starting OpenSBC

In the distrib folder under /usr/src/opensbc there should be a couple of scripts. Use ./startup.sh to start the services and ./shutdown.sh to stop them.

Access the OpenSBC management then at http://inside.ip.addr.offw:9999

Hope this helps others who have been struggling getting this going! I'm working on some Vyatta configuration files for making things all work together.
.

Sunday, December 28, 2008

Looking for an SBC solution? Vyatta + Freeswitch...

Started working on getting a proper SBC for the home network. I hacked on Vyatta with OpenSBC this morning with less than stellar compilation results. I may revisit that.

I did get a good start on Vyatta (beta of Ver 5... Debiab) with Freeswitch installed though. Here are my installation notes so far:

Vyatta VC5 Beta ISO download is at: http://www.vyatta.org/downloads

Vyatta (ver 5 beta) with FreeSwitch Installation

Install Vyatta

Boot from ISO. Press ENTER.

At login, username 'root', password 'vyatta'.

Enter 'install-system'

Configure as needed...


Get items required for building Freeswitch

Login to Firewall as user 'vyatta'

vyatta@FW1:~$ cd /etc/apt
vyatta@FW1:/etc/apt$ su
Password: (enter root password)
vyatta@FW1:/etc/apt$ nano -w sources.list

Add line: "deb ftp://ftp.us.debian.org/debian/ lenny main contrib non-free"

Ctrl-X and Y to overwrite

FW1:/etc/apt# apt-get update

FW1:/etc/apt# apt-get install -y mc autoconf automake libtool build-essential libxml2 libxml2-dev libtiff4 libtiff4-dev php5 php5-cli php5-mysql php5 php5-cli php5-mysql php5-gd mysql-server libmysqlclient15-dev php-pear php-db curl sox apache2 libssl-dev libncurses5-dev bison libaudiofile-dev subversion libnewt-dev libcurl3-dev libnet-ssleay-perl openssl ssl-cert libauthen-pam-perl libio-pty-perl libmd5-perl libpg-perl libdbd-pg-perl php5-pgsql sqlite3 libsqlite3-dev openssl ssl-cert libapache2-mod-php5 php5-cli php5-common phpMyAdmin php5-mcrypt mcrypt phppgadmin apache2 libmcrypt-dev


Get the Freeswitch Code

cd /usr/src/

svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch


Build Freeswitch...

cd freeswitch

./bootstrap.sh

./configure –prefix=/usr/local/freeswitch

make

make install

make sounds-install

make moh-install

make samples


Run Freeswitch...

cd /usr/local/freeswitch/bin

./freeswitch


That's as far as I've taken it at this point...

Shout out to http://madhawa.com/?p=10 for shedding light on Debian (etch)  install of Freeswitch.
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Saturday, December 20, 2008

New Snom 820

Looks like Snom is finally getting their act together with their phones. They've always had the technology underneath (much better from an administrative perspective than Polycom) but their packaging left a LOT to be desired.



The new Snom 820 is set to change that. With sleek new looks, finally no paper templates, an available wireless version and Snom's great software under the hood, the 820 looks ready for prime time.

Take a closer look: http://www.snom.com/sv/products/snom-820/

Let's hope they follow it up quickly with an 860 or 870 for the attendant position.
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Friday, December 12, 2008

sipXecs updates..

Version 3.10.3 Stable has hit the streets. I've done a couple upgrades at this point and all is working great.

3.10.30 combined with Polycom 3.1.1 firmware and 4.1.2 bootrom brings Music On Hold to the Polycom phones finally! Woohoo!

Oh, and they fixed a few other bugs too...

4.0 development seems to be pressing ahead quite well. The developers have just upped the development version to 3.11.9. I have been installing 3.11.8 regularly in a virtual and things seem to be shaping up quite nicely.

.

Wednesday, November 26, 2008

Audiocodes adds IP Phones to Offerings

Audiocodes which has been known for many years for their gateways has added IP Phones to their repertoire. The new phones are all HD (g.722). To me, their looks are uninspiring. If their tech support is as good as their tech support has been with their gateways I'm sure they will go nowhere... Sorry, was that sarcastic?

http://www.audiocodes.com/Content.aspx?voip=3112

Judge for yourself I guess...
.

Thursday, November 6, 2008

Been a while...

Gosh, it's been a month since I posted... doh!

October and November are usually crazy months for me and seemingly even more so this year.

Was able to get a little more bird hunting in this year which was nice and I've already bagged my deer (an hour and fifteen minutes into the season... oh well...). I've got one more autocross coming this weekend too and then am headed for the Nascar finale in Miami the next weekend.

The development builds for sipXecs 4.0 (3.11.x) are getting better and better with every passing week. Looks like the clustering pieces are finally in place and I'll be doing some testing with that soon. I'm still not exactly sure how they will be dealing with voicemail / media services in a clustered environment... hope to have that answered soon.

The concept of locations & gateway resources at locations seems to be coming together. This should allow a group of phones to prefer particular gateways based on what site they are at. I'm expecting this to be a huge benefit to a clustered system.

I've got another project I'm working on and expect to be able to provide info on that next month or in January.

.

Tuesday, October 7, 2008

Wireless DECT for SoundPoint IP Phones

Tony Graziano over at myITdepartment pointed me to a review he just did of the Jabra headset used in conjunction with Polycom 650 phones.

Recently, we decided to investigate headsets that were compatible with Polycom Soundpoint IP phones (SIP).

Polycom is kind enough to publish a known tested and certified list. As new models keep coming out though, the list is sorely out of date. One of the things we were trying NOT to do was use bluetooth headset. So we settled on a DECT 6.0 headset. After researching, we found that JABRA makes a wireless DECT headset that connects to BOTH the PC and phone simultaneously.Plantronics makes an excellent headset that connects to the phone only, but in our case we wanted a dual purpose headset and found this was about the same cost as the Plantronics with some extra functionality for the PC.

After reading up on it, we found there are different ordering items in relation to that headset. There was a mechanical lifter option (yuck), and there was also a EHS (Electronic Hook Switch) for Polycom phones (hooray!).


Read the rest of the review
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Sunday, September 21, 2008

Get MoH on Polycom Phones before 4.0 is release

A workaround was posted to the sipx-users mailing list by Chris St. Dennis. You need to have Polycom firmware 3.1.0b firmware on the phone.

To make Music on Hold work with 3.8.x or later (replace domain.com with your SIP domain), add:
"~~mh~@domain.com"/>
to the section of -sipx-config.cfg

It has been reported that this works with sipx 3.8.1 and 3.10.2 between phones and phone to AudioCodes and Patton gateways.

To modify the template so that when phone profiles are generated the settings get in there for all Polycom phones, edit the template file /etc/sipxpbx/polycom/mac-address.d/sip-2.0.cfg.vm.

Chris was kind enough to provide a patch for anyone else wanting to do this (domain is hardcoded in so change domain.com to your SIP domain, use at your own risk):
---------------- snip ---------------
--- /etc/sipxpbx/polycom/mac-address.d/sip-2.0.cfg.vm.old 2008-09-19 16:09:56.000000000 -0700
+++ /etc/sipxpbx/polycom/mac-address.d/sip-2.0.cfg.vm 2008-09-19 16:09:41.000000000 -0700
@@ -21,6 +21,7 @@
voIpProt.SIP.${setting.ProfileName}="$!{setting.Value}"
#end
>
+
<musicOnHold voIpProt.SIP.musicOnHold.uri="~~mh~@domain.com"/;>
<outboundProxy
#set ($group = $cfg.EndpointSettings.getSetting('voIpProt').getSetting('SIP.outboundProxy'))
#foreach ($setting in $cfg.getSettings($group))

---------------- snip ---------------

Add the above into a text file called polypatch in the /etc/sipxpbx/polycom/mac-address.d folder and issue the patch command:

cd /etc/sipxpbx/polycom/mac-address.d
patch -p0 <polypatch

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Saturday, September 20, 2008

sipXecs 4.0

It's been a little quiet on the sipX world lately.

Great progress is being made on version 4.0. Looks like the release however is going to be pushed to late February of 2009. It will be a big release however with NAT Traversal, ITSP bridging, Conferencing Server (thanks to FreePBX), Music on Hold for the Polycoms & their version 3.1.0 firmware and the addition of the Aastra 5x series of phones into the configurator.

Seems like the assimilation of Pingtel by Nortel has gone relatively smooth (from the outside looking in). Development is continuing at a hurried pace with the extra resouces available now as well.
.

New IP Console and Softphone Coming for sipXecs

Last year Nortel signed up Advatel from Australia as a development partner. The fruits of the partnership look like they are paying off for the sipXecs / Nortel SCS500 customers.

Advatel's PhoneEasy IP Console and PhoneEasy IP Desktop are now orderable items through Nortel for use with the SCS500. Press Release



















Additionally, earlier this year Advatel had signed up Telesource Services, LLC as a distribution partner in the US for the PhoneEasy line of products. Press Release
.

Saturday, August 30, 2008

Edgewater Networks EdgeMarc 4500 Installation Notes

Wanted to update all on the install of the EdgeMarc 4500 devices for our customer last week.

Getting into the units was pretty easy. Power them up, plug into LAN port 4. The unit is setup with DHCP enabled and configured for 192.168.1.1. Fire up the web browser and point it at that IP. Default login is root with a password of default. I changed the IP which caused the unit to reboot (a little to my surprise). The VLan enable check box is on this page, if you enable VLans, the only port that is not a trunk port is LAN port 4.

When changing the IP, the EdgeMarc does not update the default DHCP table. You'll need to statically setup an IP in the new address range you just setup for the new LAN IP. One piece on the DHCP page that is missing is the ability to pass a DNS domain name in the DHCP scope. I'd really like to see this changed as DNS is so important to the sipX world.

I configured VLans 1 (data) and 2 (voice) at each site. Put LAN ports 1, 3 and 4 into VLan 1 and port 2 in VLan 2. I setup the 3 devices we got in a fully meshed VPN configuration. To get ports statically into a particular VLan the port must be configured for 802.1 instead of 802.1q (trunk port).

Routing between VLans is enabled by default. I didn't try to block any traffic between VLans for this application.

In defining tunnels I could only specify one to one subnets (unlike on a Cisco ASA or PIX). So I had to create a tunnel for each VLan and each site (Data to Data, Phone to Phone, Data to Phone and Phone to Data). What a pain... but it works. With 3 sites fully meshed I ended up with 8 VPN definitions at each site.

Once the basic configs were done I went into the QOS setups and prioritized traffic from each of the Voip cards outbound.

The only gotchas for me with these units are the reboots after many of the changes and the lack of a DNS Domain option in the DHCP configuration. Looking forward to testing these as a SIP failover device.

.

Saturday, August 23, 2008

Edgewater Networks EdgeMarc

Finally got my hands on a couple of Edgewater Networks EdgeMarc devices yesterday. I have to say, they really seem like the 'magic bullet' for almost any kind of Internet based VoIP solution.

Highlights:
  • Build in Firewall
  • VPN capabilities
  • Traffic Shaper (by ToS, DSCP, IP Address)
  • WAN Router with up to 4 T1's (MLPPP support)
  • FXO / FXS Gateway
  • 4 Port Network Switch
  • 802.1Q VLan Support
  • Call Quality Monitoring device
  • Local NAT / DHCP
  • Local TFTP and FTP Server
  • Optional Site Survivability
  • SIP NAT Traversal & Application Layer Gateway
The units I got were 4500 series for a particular project and don't have the FXO / FXS ports so I am unable to test that functionality with sipX. In this particular project I need to interconnect 3 sites that have traditional PBX's with proprietary VoIP trunk cards in them, prioritize that traffic over everything else and mesh VPN the sites. I plan on creating a separate VLan for the VoIP cards and plugging the VoIP card directly into one of the switch ports on the EdgeMarc, and prioritizing all of that IP address' traffic outbound.

The site survivability for SIP is an optional component that must be licensed. It monitors the availability of a PBX (via various methods including Ping) and if that connection is not present can re-route calls out a PSTN connection connected to the unit or to a gateway at that site.

The firewall pages could use a little 'dressing up' for creating rules and NAT translations. The GUI management is not as slick as a typical firewall device.

I'll planning on getting in a couple units for testing with some FX ports... when then 'crazy season' ends.

.

Friday, August 15, 2008

Comments around the web on Nortel's Acquisition of Pingtel

The industry seems to be all a twitter (pun intended) about Nortel buying Pingtel. I'm really hoping they don't mess up a good thing with the acquision. I'm encouraged by the cross pollination that has been going on for about a year with the developers working together.

Ken Camp blogged about the Pingtel acquisition today... Ken always brings a unique perspective on things.

Raltime Unified Communications Blog

Over at OS-Voip they feel it is a big day for Open Source Voip. For sipX the real boost came about a year ago when Nortel started contributing to the development. I really think they were just doing their due dilligence.

OS-Voip Blog

Over at the Hyperconnected Enterprise, Tony Rybczynsk of Nortel writes briefly about the acquisition.

The Hyperconnected Enterprise Blog

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Thursday, August 14, 2008

Nortel purchases Pingtel Assets from Bluesocket

This just seems like a better fit for Pingtel rather than being with Bluesocket... That's not meant to be a knock on Bluesocket...

Here's the press release from Nortel's site: Press Release


Scott Lawrence from Pingtel posted this to sipX-users:

Effective last Monday (11 Aug), the assets of the former Pingtel have been purchased by Nortel Networks Corporation [1]. As most members of our community will have noticed, Nortel has been a major supporter of the sipXecs project for over a year now, with many Nortel employees making important contributions. The Nortel SCS 500 http://www.nortel.com/scs500 is based on sipXecs.

I'm happy to say that the entire Pingtel team has moved to Nortel, and that our commitment to the sipXecs open source project is unchanged. The project goes on as it has been; indeed, with the greater resources of Nortel behind it, we expect to be able to expand our scope considerably.

This will not affect any of the project infrastructure - all the lists and other resources remain the same. There will be some brief outages in some services when we move some of the servers to a Nortel facility, but these should not last more than a few days. You'll see the pingtel.com email addresses change to nortel.com addresses.

Having this kind of commitment from an industry leader in telecom is a great endorsement of all that the sipXecs community has achieved. To all of you who have contributed your ideas, your enthusiasm, and your code: thank you again, and I hope you'll join me in celebrating this new milestone.

Saturday, August 9, 2008

SIP NAT Traversal

So, you've got your fancy SIP phone system all setup and now you want to connect it to the outside world (to an ITSP or to another SIP PBX). Some may be lucky enough to have NAT traversal built in but many are not. Even if you do have NAT traversal built into your product, how secure is it? Does it support near-end and far-end NAT traversal?

What is the problem with SIP and NAT? It was detailed very well at freshmeat in a tutorial. NAT Traversal for the SIP protocol

In the sipX world we'll need to wait for release 4.0 for NAT traversal to be built in.

In the mean time here are some possible solutions.

Ingate - Ingate's Siparator or SIP firewalls are secure and reliable means of getting near and far-end NAT traversal. The remote NAT traversal is an option that allows for configuration-free (client side) NAT traversal or allows you to set it up as a STUN server. I've used Ingate's products many times and am very happy with their support and configuration options. www.ingate.com

Edgewater Networks - I haven't had a chance to work with these products yet but their Edgemark line of equipment looks very promising for Internet edge connectivity. www.edgewaternetworks.com

OpenSBC - Seems to be one of the more popular choices in the open source world. I haven't tried it myself as I've really needed some commercially supported solution like the Ingate. If somebody has tried it please comment... http://www.opensipstack.org/

Others people like?

Tuesday, August 5, 2008

Update on Polycom Firmware

Got an e-mail from a reader who talked with Polycom engineering. Looks like 3.1.0 is 6 weeks out... So it's looking like middle of September.

Saturday, August 2, 2008

New Polycom Firmware on the way

Polycom SIP firmware version 3.1.0 is due around the end of August. We knew it was coming as Pingtel said they were testing an engineering release, we just didn't know when... and now we do!

The big news for this firmware is Polycom finally supporting the Music on Hold (MoH) standard that sipX utilizes (the Dale Worley IETF Draft)

This will be a huge boost for users of the system!

At present the Snom phones are one of the few who support MoH on sipX.

Here's the sipX MoH page: MoH Page

Thursday, July 31, 2008

Product selection, product selection, product selection...

One of the most important decisions that must be made early on in an implementation is what products to utilize.

Traditional PBX vendors have been able to dictate which hard and soft phones you have to use and thus are able to control feature set.

Part of the bugaboo of a standards based solution is that all products do not support all of the standards. You can expect that all products will not have equal functionality. Even if a vendor does support the standards you are looking for, they may have errors in their code or they may have interpreted the standards differently than other vendors.

Testing is critical to the process. Make sure the equipment you select will match the requirements you have.

So, what are your requirements? Low cost, certain features?

Wednesday, July 30, 2008

Asterisk may be older, but sipXecs is better

Found this interesting blog entry from Tony Rybczynsk of Nortel...

Asterisk may be older, but sipXecs is better

You have probably read about Nortel's Software Communications System 500 (SCS500), a Unified Communications (UC) SIP-centric software solution for SMB (30-500 users), and that our go to market includes IBM and Dell.

What you may not know is that the SCS500 is based on open source from SIPfoundry, and blends the best of both the open source framework and Nortel's experience and expertise in voice, data, multimedia and unified communications. Why did we choose sipXecs from SIPfoundry as the basis for SCS500?

Read More


Linksys SPA941 and sipX

Got a chance to work with some Linksys SPA941 phones and sipX.

These phones were connected to an Asterisk system originally and we were moving the customer over to sipX due to a system failure.

Ran into a few problems with the phones... First thing I did was a factory reset on the phones (needed to be done right on each phone in the setup menu).

I upgraded each of the phones individually with the most current firmware. I could not find good information as to how to automate it so i used the manual method (http://ip.of.phone/upgrade?tftp://ip.of.sipx.pbx/spa.bin). I did use the sipX device file manager to load the .bin file to the pbx.

The next challenge was getting the phone to pickup the config generated by sipXconfig. Finally figured out to go to the Admin / Advanced / Provisioning page on each phone and change the profile rule to read '/spa$MA.cfg'.

Still having issues with call pickup from park orbit and attended transfer of calls.

I created a new wiki page documenting what I had found: Linksys sipX Configuration

Tuesday, July 29, 2008

3.10.2 Update

Version 3.10.2 was released on July 23rd.

There were several bug fixes as well as repairing some upgrade issues from version 3.8 to 3.10.

I had done two 3.8 to 3.10.1 upgrades but had been holding off others due to reported problems (not that I experienced any). Now that 3.10.2 is out it's time to get systems upgraded. I believe most of the problems had centered around HA systems.

3.10.x provides some great call routing capabilities that we had to rely on gateways for in the past. This simplifies installs and allows the end-user an easier way of tweaking their routes.

I have done an upgrade from 3.10.1 to 3.10.2 on my test system and also done a fresh install of 3.10.2 from ISO (http://sipxecs.sipfoundry.org/pub/sipXecs/ISO/sipfoundry-3.10.2-centos5-i386.iso) without problems.

New Blog

I had kind of fallen off the face of the blogging world... trying to get back into the habit with something a little more focused around a product I spend a lot of time with... sipXecs.

sipXecs is "the other" open source PBX available to the masses. It takes a decidedly different approach to an office communications system.

I'll be posting product news, installation tips and tricks as well as different equipment I've tested and fought with.