I am sabarinath using ALU ip touch 4018 EE. I want to use the phone with asterisk PBX for a test purpose. My NOE is 4.11.00. I want to change that to SIP.
Please provide me a step by step procedure to manually or provide me a proper document
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This PBX system are you using? if it is an asterisk based pbx, you will need to do a GotoIf statement in extensions.conf.
If you are using UI based system, you can simply do it in the timegroup and timecondition page.
I have a similar phone, the IP8830. Press the button to the left of the round button (looks like a cog) . This is the configuration menu key. Press 1 then 2 and get your IP. Once you have your IP, enter it in your web browser so it looks like this (http://192.168.1.200:8000) . The default user name on mine is "private" and the default password is "lip" . Once you are logged in you can go to "Phone Settings" and set the number of lines on the keys. Next you can go to "Programmable Keys" to set your BLF and Speed Dial Keys. I am using SIP firmware with a Asterisk PBX so yours may be different. My firmware version is "1.1.04spbx_a" .
IP Phones do not provide voicemail, that is a function of the PBX it is connected to, either a premise IP PBX or a cloud-based (hosted) PBX. If you are using an Asterisk-based PBX system, the usual way to access the voicemail system is to dial 98 from the phone, however you will need to know the voicemail password.
3Com 3100 series phones do not support SIP with non-3Com systems. They have a basic boot loader which must download code from a 3Com NBX or a 3Com VCX system. If you don't have either of these, then you won't get runtime code on the phone, thereby making it impossible to use the thing with Asterisk.
Interesting question. Thats what today call centers are doing with predictive dialer solution.
But if you dont want to go that much advanced, you can opt for CRM connectors. In general they are called CTI. Computer telephony integration. For Mitel IP Phone, I dont know what PBX you are using. Refer : http://www.mitel.com/resources/ap_51009460RA-EN.pdf The following are general steps. 1. Create CTI account in your PBX. 2. Download a CTI client. 3. Connect CTI client with your PBX and map with your extension. 4. Whenever you get a call a page will open. Opening a link when no answer is depends on CTI client.
The exact method really depends on your ip-pbx and whether or not you will have a persistent VPN between the remote site and the main site.
If you have the VPN then the most you would have to do different would likely be to manually set the ip settings for the phone.
If not, then you will likely be dealing with a pretty complex implementation that will require in depth work with the PBX, firewall, DNS, and possibly security certificates.
Some IP-PBX units do decent job of shielding you from the gory details but without more info to go on, i.e. details on the PBX type and whether or not you will be using a VPN, this is about as far as we can go at the moment.
If you need more, please post a follow up with additional information and we'll be glad to continue.