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A customers network and his Virtual PBX!
Most the time we get new customers they configure their PBXs and we do not here from them again! Other times tickets go back and forth for days. We get the occasional ticket from existing client here and there but mostly new clients put in the most tickets.
Recently we got a new client, we installed the server with Trixbox upgraded it and handed the Virtual PBX off to the customer. They were having all kinds of problems and did not understand why. So we installed Elastix. At first they said they were having no issues and it was working. We resolved the ticket. Later on that day they were getting disconnected from the PBX phones would go offline and then come back online. We troubleshot the Virtual PBX from our end and did not see any connectivity issues, hardware issues, or software problems beyond configuration. The customer was getting very frustrated stating that it worked at their office fine etc. Then she had us install PBXinaFlash. Everything was working great for about six hours and BAM there was another ticket! We again asked a hundred questions and got very little answers. Finally the customer told us every time we reinstalled she had rebooted his router. She rebooted it again and the phones came back online. Keep in mind every time we did traceroutes to his network and his provider everything looked great from his Virtual PBX. His phone would connect then just show offline. She ended up buying a new router that was VOIP compatible. We could not ever get her to do a ping or traceroute to her virtual server. We left PBXinaFlash on the server as by default it installed fail2ban and iptables. She needs security I do not think she knows how to ssh into her virtual server but she can use the web interface webmin, freepbx etc!

Recently we had a client that wanted an email solution along with a CRM solutions. We looked around and finally decided to give Zimbra a try. It was easy to install on a CentOS vps. Basically you get the server ready and run the install script. We got it up in a few hours and tested it. Zimbra seemed very easy to use and setup. The next challenge was getting up a CRM installed. We tried to use the Zimbra SQL server and Web Server but that seemed to be to much work and might cause problems later on. We installed another MySQL Server and Apache server which was easy as Zimbra uses different ports for both MySQL and Apache. We went ahead and installed VTigerCRM. We tested everything and it worked pretty well. We handed it off to the client to test. They loved the solution. All in all Zimbra seems to be a great mail solution. It even seems to be a good replacement for Microsoft Exchange. VTiger is also great CRM…

Recently on one of the Elastix PBXs I host, I had to put up another internal network. Basically two sites were VPNed together so on the other end of the VPN were the new phones were there was another internal network. When someone would call out they could hear me but I could not hear them. This was caused by the sip_nat.conf not having the new internal network. This can also happen if you do not have the internal network and you are using a sip provider.

You have to setup the sip_nat.conf with the External IP and the Internal Network.

externip=123.123.123.123
localnet=192.168.0.0/255.255.255.0

They have 3 different networks so they have three entries for the localnet.

Here is what we did to make it work.

localnet=192.168.0.0/255.255.255.0
localnet=172.16.0.0/255.255.0.0
localnet=192.21.10.0/255.255.255.0

Once we did this all calls worked on all three networks…

Has Trixbox development stopped? Should you look for other Asterisk solutions?

Has Trixbox development stopped? Well I am not sure if it has stopped completely. For version 2.6 yes it does seem like it has stopped. Here are the most recent release dates.

Trixbox 2.8          latest release 2010-06-11

Trixbox 2.6          latest release 2009-06-22

We have a lot of customer that like Trixbox. We also have seen a lot of Trixbox servers get hacked in the past. Always update your server and patch exploits. Use APF, BFD, and Fail2Ban if you can. We recommend locking down your PBX (Trixbox, Elastix, or PBXinaFlash Server) to just the IP Addresses that need it. The IP Address of where your phones are connecting from, The IP Address of your VOIP Providers and the IP address of the Management Team managing your PBX (Trixbox, Elastix, or PBXinaFlash Server). One exploit that was brought to our attention recently in Trixbox is listed below.

The admin web interface can be accessed using the username of “wwwadmin” and the default Trixbox password, giving full administration privileges to the server including access to the extension and trunks settings. The versions of Trixbox affected are Trixbox CE 2.8.0.4 and below +  Trixbox CE 2.6.2.3 and below.

 We suggest that you fix this ASAP or completely firewall your Trixbox server from outside access.

You can remove the wwwadmin user completely from two files and use passwd-maint to change the default maint password which is password.

Secure your PBX and also take the time to update to the latest available version of Trixbox.

In order to secure your PBX remove the user “wwwadmin” from the following locations:

Edit the file /usr/local/apache/passwd/wwwpasswd using nano or vi from the command line

remove the line in

/usr/local/apache/passwd/wwwpasswd

that looks like

wwwadmin:40ig27asm87

Edit the file /etc/trixbox/httpdconf/trixbox.conf using nano or vi from the command line

remove wwwadmin from one line in

/etc/trixbox/httpdconf/trixbox.conf

Require user maint wwwadmin

You may find this wwwadmin user in both of these files. /usr/local/apache/passwd/wwwpasswd /etc/trixbox/httpdconf/trixbox.conf

Run the command passwd-maint to change the default maint password for accessing trixbox.

You will need to have access to the command line of your server locally or remotely using SSH and be able to login as the root user to complete this task.

We have been using Elastix since it came out. We have deployed up to 160 Phone on Elastix and Trixbox. Elastix gets the needed attention and has an updated feature set. A lot of people like PBX in a Flash as well.  It is really preference as they are all originally based off FreePBX. Trixbox came from Asterisk@Home which was also based off FreePBX. Trixbox did fork into pbxconfig but don’t forget where they came from! Elastix has an unembedded version of FreePBX which lets you use all FreePBX’s features. Elastix also has new features that are only found in Elastix like faxing and many add-ons!

Any distribution could get exploited it is important to run security updates and lock your server down as much as possible.

IPhone verse HTC Sensation (Android).

I went and bought my IPhone when the came out. I went through a few of them then the IPhone 3G came out and I eventually upgraded.  I had this IPhone for a few years. I had all of these phones unlocked on T-Mobile as I felt AT&T was overpriced. I never cared about the speeds I just needed a phone and email most the time. When there was no Internet it was nice to be able to tether and get on the net even if it was like dialup! When the IOS4 came out I upgraded. Ever since once I install to many APPS the phone is slow. My wife and I have always had a shared plan on T-Mobile. For about the last year or so I have been thinking about getting a new phone. Finally my wives phone broke recently and I had no choice but to go look at phones.

First we went to T-Mobile and looked at their Android phones. The T-Mobile guys all said get the HTC Sensation. It looked nice and it had dual processors like I had read about. They cost too much and we were not up for an upgrade. Plus my wife and I both wanted IPhones.  We went to Verizon. They had IPhone 4s for a decent price $199 for a 16 GB. By the time we got out of our current plan and bought new phones got the same features we have on our current plan it was way to much to switch to Verizon. The monthly cost were going to be around $40 more. We left Verizon and went back to T-Mobile. My wife had been talking to T-Mobile on the phone the whole time we were in Verizon and they were offering us a much better deal on the phones now. We got the HTC Sensation’s for right around $500 for two of them. Our bill stayed the same, which was cheaper than AT&T and Verizon with Unlimited everything. Well I guess DATA is not unlimited even though it says it is. They give us 5GB of High Speed transfer, which I guess is 4G.  Their site says: Unlimited data – 5 GB of high-speed data included.  The Internet is way faster than before being able to use 4G obviously.

What I really wrote this article about was the performance of the HTC sensation compared to the IPhone. It is really snappy in everything I do.  I did have an IPhone 3G so IOS4 ran slower then on an IPhone 4 but the HTC Sensation was an awesome upgrade. Almost everything I could ever want came on the phone. Tethering, SIP, Internet Email, and much more. The first thing I did was go see if the Market place had all the apps I used before. Guess what it did and they were all free! Angry Birds all three versions free! They play ads every ten minutes or something. In the past I have had about 5 different android phones given to me. I never liked them I always went back to my IPhone. I think now my IPhone will be used as a SIP Phone for about calls from the Office! The HTC has made it as an IPhone replacement for me!

I am sure the IPhone 5 will come out and have a lot of new really awesome features but this way I pay $40 a month less than AT&T or Verizon!

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Let Pinellas Hosting take care of all your Hosting needs. Dedicated Servers, Virtual Servers, VOIP, Dedicated PBX, Virtual PBX, Web Hosting, Email Hosting, DNS Hosting, Managed Services and more. Contact us for your custom quote today.

We have many services you can offer your customers here is a list.

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Dovecot and Cpanel issue!

We setup a Cpanel server recently then we started to put customers on it. Right away we had a customer tell us they were not able to get email. They were getting the following error when checking email from a help desk.

SECURITY PROBLEM: insecure server advertised AUTH=PLAIN

Retrying PLAIN authentication after Authentication failed.

First thing that came to thought was that it is using Plain Text Authentication but it is not working for whatever reason. We tried to setup email accounts in outlook and they could send but not receive. POP3 was not working. I googled around and found someone with the same issue. We played with the server a little but then switched to the old default email server in Cpanel, Courier. Once we did that all outlook accounts and the support desk started working immediately. I do not know if this is always broke but it sure wasted about 2 hours of my time.  I think I will stick with Courier  in the future.

Cannot dial anything past 211!

At the one of our customer’s sites they have extensions in the 2100 range. I did the PBX install and left. Everything seemed fine. They started dialing 2111 2112 2113 2114 and they would get all circuits are busy. The PBX was only seeing 211.

Here is what I was seeing from the Asterisk console.

[Jul 11 10:09:47] VERBOSE[3220] logger.c:   == Spawn extension (from-internal, 211, 4) exited non-zero on ‘SIP/2155-0901

[Jul 11 11:11:06] VERBOSE[3367] logger.c:     — Executing [211@from-internal:1] ResetCDR(“SIP/2155-b7881038″, “”) in ne

[Jul 11 11:11:06] VERBOSE[3367] logger.c:     — Executing [211@from-internal:2] NoCDR(“SIP/2155-b7881038″, “”) in new s

The problem ended up being the Polycoms digitmap.  We upgraded our polycoms to the latest firmware SoundPoint IP, SoundStation IP and Polycom VVX SIP 3.2.5 [Combined]  we have polycom 330, 650, 600, 4000, and 6000. Here is the default digitmap for this firmware version.

<digitmap dialplan.digitmap=”[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT” dialplan.digitmap.timeOut=”3|3|3|3|3|3″ />

The problem was this [2-9]11 for dialing like 411 and 911. I had to change it to [3-9]11 which fixed the problem. I was able to figure out the problem by reading this page http://apetec.com/voip/Polycom-DialPlan.htm

At the bottom of the page it has the following which helped me figure it out.

It means the following:

• [2-9]11: 911 rule: 211, 311, 411, 511, 611, 711, 811, 911 are dialled immediately
• 0T: Local operator rule: After dialing “0″ the phone waits T seconds and then completes the call
• 100: Auto-attendant default extension
• 101: Voicemail default extension
• 011xxx.T: International rule without prefix
• 9011xxx.T: International rule with prefix
• 1[2-9]xxxxxxxxx: LD rule without prefix
• 91[2-9]xxxxxxxxx: LD rule with prefix
• 9[2-9]xxxxxx: Local call with prefix
• *xx: 2-digit star codes
• *xx[2-7]xxx: Enables immediate dial for services like intercom/paging and directed call pickup.
• *4[2-7]xxx: Retrieve call from park (if park slot is x digit extension)
• [1-7]xx: A regular 3 digit extension that does not start with 9 or 8 is dialed immediately
• [2-7]xx: A regular 3 digit extension that does not start with 9 or 8 or 1 is dialed immediately
• [2-7]xxx: A regular 4 digit extension that does not start with 9 or 8 or 1 is dialed immediately
• [8]xxx: A 3 digit extension prefixed with an 8 (routes calls directly to voicemail of extension xxx)
• [8]xxxx: A 4 digit extension prefixed with an 8 (routes calls directly to voicemail of extension xxxx)

Here is what I saw after I fixed the issue.

[Jul 11 16:16:15] VERBOSE[4448] logger.c:     — Executing [2118@from-internal:1] Macro(“SIP/2155-090c0ba0″, “exten-vm|2

[Jul 11 16:16:15] VERBOSE[4448] logger.c:     — Executing [s@macro-user-callerid:13] NoOp(“SIP/2155-090c0ba0″, “TTL:  A

[Jul 11 16:16:15] VERBOSE[4448] logger.c:     — Executing [s@macro-exten-vm:3] Set(“SIP/2155-090c0ba0″, “VMBOX=2118″) i

[Jul 11 16:16:15] VERBOSE[4448] logger.c:     – Executing [s@macro-exten-vm:4] Set(“SIP/2155-090c0ba0″, “EXTTOCALL=2118

[Jul 11 16:16:15] VERBOSE[4448] logger.c:     — Executing [s@macro-exten-vm:8] Macro(“SIP/2155-090c0ba0″, “record-enabl

[Jul 11 16:16:15] VERBOSE[4448] logger.c:     — Executing [s@macro-exten-vm:9] Macro(“SIP/2155-090c0ba0″, “dial|20|tr|2

[Jul 11 16:16:15] VERBOSE[4448] logger.c:     –  dialparties.agi: dbset CALLTRACE/2118 to 2155

[Jul 11 16:16:15] VERBOSE[4448] logger.c:     — Executing [s@macro-dial:7] Dial(“SIP/2155-090c0ba0″, “SIP/2118|20|tr”

PBX Move
Moving an Asterisk PBX can be pretty easy. There are some things that you should look at. We recently moved a trixbox to a new subnet in a new building. One of the problems was no sound on calls. A lot of time this can be caused by incorrect configuration in the sip_nat.conf in /etc/asterisk once we added the new ips the sound started working. We also decided to use tftp as we have to roll out about 70 new phones. We always download the newest firmware and sip files. On polycoms you download the new files throw them into /tftpboot then you change the polycoms from FTP to TrivialFTP add the PBX ip as the server and bam your phone will grab the configs off the PBX. This particular PBX is trixbox. We used endpoint manager to make all the config files. You will want to make sure you have a server.cfg with the correct information. This is what we have in our server.cfg

<?xml version=”1.0″ encoding=”UTF-8″ standalone=”yes”?>
<localcfg>
<server voIpProt.server.1.address=”172.16.0.100″/>
<SIP>
<outboundProxy voIpProt.SIP.outboundProxy.address=”172.16.0.100″/>
</SIP>
<SNTP
tcpIpApp.sntp.daylightSavings.enable=”1″
tcpIpApp.sntp.address=”172.16.0.100″
tcpIpApp.sntp.gmtOffset=”-18000″
/>
<localcfg>