A customer networks and his Virtual PBX!
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!
Changing Internal networks or adding another internal network in Elastix, Trixbox, or FreePBX
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 like Elastix or PBXinaFlash?
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.
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Cannot dial anything past 211!
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”