Knowledge base VoIP setup
Firewall Settings
This is what we suggest firewall-wise for voip customers:
- Avoid using NAT where possible.
- Allow SIP from our servers: UDP port 5060 and 2002 from 81.187.30.110 - 81.187.30.119 (we may add more servers in the future, but will be listed here and the status page when we do)
- Allow RTP from everywhere: UDP 1024-65535* from everywhere (which also covers SIP above)
SIP is the call routing information that creates and manages calls
RTP is the actual audio, as it will be as direct as possible the audio can be sent from anywhere on the internet.
*on most phones you can configure which ports to use for RTP, so you can restrict this range further
eg on a SNOM phone the default range for RTP is 49152 to 65534.
SIP settings
When setting up SIP you need to know your sip number, e.g. 01234567890, that we have allocated to you, and your password. You will have been sent these when you requested the number.
The other bit of information you need is the name of the call server. In different sip devices there are different names. In some cases you may have to enter a proxy server and in some cases a registration server and in some cases both. The server is your sip number followed by .call.me.uk, e.g. 01234567890.call.me.uk
If you have to enter something that looks like an email address, or has sip: at the front it probably needs your number, an @, and then your server, e.g. 01234567890@0123456789.call.me.uk
If you are asked for a realm, just put asterisk
Control pages
Our control pages allow a number of options for each number.
| ACR | Anonymous call reject - callers hiding their number are rejected with a suitable message. |
| Q | This is a call queue so callers wait rather than getting busy, if possible. |
| Record | Defines if you want calls recorded or not. Set the Email address first. The recordings are sent to the specified email. You can also select if you want calls encoded as MP3 or OGG format. |
| Warn | This is an experimental feature which will give a warning when the call is connected advising the call is being recorded. |
| VM | Defines the voicemail settings. |
| Fail | A number to divert the call to if the call fails for some reason (e.g. destination is not reachable). This must be a normal phone number, no spaces. |
| Also ring | Up to 10 numbers that are also rang when the call is received. Normal phone numbers, no spaces. If you prefix * that delays ringing. Prefix ** delays more. Prefix ! to delay ring all but one of those prefixed ! (randomly). |
| Incoming calls | You can define profiles for times of day and day of week and select one of these for incoming calls. You can also pick a number to divert to outside these hours. |
In addition to these you can set your email address, and email for incoming SMS (where available), the password for SIP login, and how you would like the call delivered. The usual option is simply a SIP phone, but you can have a call delivered to your phone system using IAX, SIP, or H323. You can also force the call to voicemail or have it answered as a fax which is emailed to you. Note that fax service is experimental.
Test Numbers
We have a few numbers for test purposes that you can dial - there is no charge for calling these:- 17070 - Reads back your login and your caller id
- 17071 - miliwatt - continuous tone, useful to test for jitter
- 17072 - Speaking clock
- 17073 - Readsback your caller ID
- 17074 - Echo test, useful to test latency
- 17075 - DTMF test, type a number and it will say it back
