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...
.
Wednesday, November 26, 2008
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.
.
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.
Read the rest of the review
.
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
.
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
.
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.
.
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
.
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.
.
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:
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.
.
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 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
.
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
.
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:
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.
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