I am going to start off by saying, I am a complete noob when in comes to this technology. A few months ago I got put in charge of a new project. Part of the project is integration of a phone system with a video conferencing system. I have the video conferencing system up and running and just started on the phone system. To save money, we have decided to go with an asterisk server and a SIP Trunk for our phone system. I used AsteriskNow 1.7.0 32bit.
Details:
FreePBX 2.7.0.5
CentOS 5
Asterisk 1.6
I will set up a zone in my VCS Control that will route calls to the FreePBX server, but the part I am having trouble with is a SIP trunk on the FreePBX server…
To register to my VCS Control, not much is needed. It is open registration, so no password is needed, just a username. It is located on the same internal network. I have tried playing around with a bunch of different settings in the SIP Trunk, but haven’t been able to register with any of it.
This is a copy of settings I used to register with a different device:
SIP Registrar Domain: 192.168.1.36
SIP Registrar Type: Standard SIP
Username: Codian
Password: is left blank
SIP Proxy Address: 192.168.1.36
Outgoing Transport: UDP
That is all I need to specify to register with another device… so I hope that helps.
I haven’t found any internal SIP Trunks that don’t require a password. Maybe I am approaching it wrong? I am able to register devices independently to my FreePBX Server as extensions. What are the channel (simultaneaous call) restrictions on a single extension?
Tips, help, and constructive criticism wholeheartedly welcomed.
Good thing I was subscribed to this thread! I haven’t logged in here for a while.
VCS Setup:
Create a Neighbor Zone for Asterisk on Port 5060 via UDP.
Create a Search Rule(Here is the rule I setup):
RULE NAME: PBX SEARCH
DESCRIPTION: TRANSFER CALLS TO PBX
PRIORITY: 600 (This is that last rule)
SOURCE: ANY
MODE: ALIASPATTERNMATCH
PATTERN TYPE: REGEX
PATTERN STRING: (.*)@.+
PATTERN BEHAVIOR: REPLACE
REPLACE STRING: \[email protected] (Just change this to your asterisk IP)
ON SUCCESSFUL MATCH: CONTINUE
TARGET ZONE: ASTERISK
STATE: ENABLED
This enabled me to pass calls to the Asterisk Server from my VCS. Since the VCS never changed I could add call handling in the Asterisk Server to forward the call to my SIP Trunk provider.
Asterisk Setup:
Note, most of the setup ended up being via telnet. I didn’t use FreePBX since I couldn’t find help. Also, I tested a lot of things so some of this might not apply… its been a while so I don’t remember specifics.
I also I have the information for my SIP Trunk provider (8x8) in this file.
In the extensions_additional.conf I have:
[from-trunk-sip-CodianTrunk]
TRUNK=SIP/packet8
exten => _1.,1,DIAL(SIP/${EXTEN}@packet8,r)
; end of [from-trunk-sip-CodianTrunk]
So, any call coming from my VCS (I was orginally having it registered to the Codian, hence the name) will be routed directly to my SIP Trunk provider with their specified call headings.
Took me about a week to figure this one out, hopefully your company can provide their own copper lines or has a provider that is a bit more knowledgeable. Probably not the best, but it works, and I have no desire to change it since none of the calls from my VCS need to be calling other extensions.
I’d love to talk to you sometime about how you are using it. Feel free to contact me at my google account. My name without the space. Any solution though, please post here.