How to configure a polycom 501 SIP with asterisks/PBX in a flash through the web GUI
How to configure a polycom 501 SIP with asterisks/PBX in a flash through the web GUI
First thing you will need to do is get the IP address of the unit. To do this click Menu then 2,2,1
Once you have your IP web into it
The default username is “Polycom” case sensitive and the password is “456” both without the quotes.
If something isn’t mentioned to fill in just leave the defaults
Next
Click on Lines
- Display Name = your extension number
- Address = ext number
- Auth User ID = ext number
- Auth Password = password that you set
- Label = ext number
Number of Line Keys = 1 (in most cases)
Server 1
- Address = IP or hostname of phone system (asterisk server)
- Port = 5060
- Transport = DNSnaptr
- Expires = blank, or if necessary, 30
- Register =1
- Retry time out = blank
- Retry Max Count = blank
- Line Sieze Time Out = 30
You can configure server 2 if you have a fail over server
Message Center
this section configures the messages button on your phone. This is nice so when you click it your messages will be checked.
Subscriber = blank
- Callback Mode = Contact
- Callback Contact = *97 (this is the default set in asterisks unless you changed it)
You can configure the other lines 2 and 3 if you want to with different extensions.
Click on submit (this is irritating as the phone reboots and takes a while to do so)
Click on SIP
Under Outbound Proxy
Address = blank
Port = blank
Transport = DNSnaptr
Server 1
Address = blank
Port = blank
Leave the rest at factory settings, but scroll down to…
Local Settings
Local SIP Port = (if this is a remote phone, this must be a unique number for each phone on the network, i.e.5060, 5061, etc. If a single phone, or on the same network as the PIAF, leave blank or set to 5060) also make sure that you have this ports open on your firewall going out, they are UDP.
- Digitmap = [2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[1-8]xxT
(this is what I used, but google around to know how it works and what this is. Basically this tells your phone how to dial, when to wait, etc. above is setup for a three digit dialing (at the end))
- Digitmap Timeout = 3|3|3|3|3|3
- Digitmap Impossible Match = 0
Click on Submit. Let the phone reboot and login again
Click on Network.
RTP Port Settings
- Forced Port = blank
- Port Range Start = blank
Network Address Translation
- IP Address = blank
- Signalling Port = blank
- Media Port Start = blank
Click on submit and let phone reboot
Click on the General tab
User Preferences
- One Touch Voice Mail = enable
Time
- SNTP Server = ntp.nasa.gov (or whatever)
- GMT Offset = (-8 here in California)
- SNTP Resync Period = 86400
Set the Daylight Savings for your needs. Polycom has a word on this- Google for the correct settings, else your time may be off for part of the year.
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Hello! Finally I have found help to configure my Elaxtix PBX server with 2- Polycom 501 and 1-Polycom 550 sip phones. Have had the setup for 2 years and never successfully implemented it, gave up frustrated. The instruction you gave here got it up and running, able to call my other two extensions, call out and in! I still have setup issues… don’t know how to setup inbound to ivr, weird receptionist voice with “leave a message”….. I don’t know if she is in Asterix or in my 501 polycom. Could I get further help if needed? Thanks so much!