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Category:VoIP Faults: Difference between revisions
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<indicator name="VoIP">[[File:menu-voip.svg|link=:Category:VoIP|30px|Back up to the VoIP and SMS Category Page]]</indicator>
<indicator name="Faults">[[File:Main-fault.svg|link=:Category:Faults|30px|Back up to the Faults Category]]</indicator>
With most VoIP problems it is best to contact Support. However, here are a few pointers:
==
===Unable to register===
[[File:Snom-notregistered.png|right|frame|The screen of a Snom]]
*Check username and password carefully. (Typically the username is the full phone number in international format, eg +44.....)
*See the [[VoIP_Phones_-_Generic_Client|General VoIP settings]] page for registration details
*See the [[:Category:VoIP Phones|VoIP clients page]] to see if your VoIP phone is listed and has any special notes
*Check if your router has a SIP ALG - see: [[Disable_SIP_ALG]]
=== Only some calls are getting through ===
==One-way audio==▼
This is a common issue that can occur when you are using multiple phone in a 'natted' environment. We have an article [[VoIP_NAT|here]] on this particular issue.
Our support team can also assign you a small block of IPs if necessary.
=== Using an ATA but the phone doesn't actually ring, but it still works ===
If you have an older phone plugged in to a VoIP-to-Analogue ATA, and if calls work, but the phone doesn't actually ring, then you may be needing a RG11-to-BT adapter that includes a ring capacitor, such as https://www.broadbandbuyer.com/products/2370-esscable-rjp-bts/
=== Ghost calls /phantom calls / Strange CLI, nobody there===
If you receive phantom calls, perhaps from a CLI (eg '1001' or 'test' or '2000001' or '0700' etc) that is odd looking, and when you answer, there's no one then these are probably from portscans of your network and not actually from the A&A VoIP servers - they won't appear on the Call Log records.
These scans result in your phone ringing with a strange caller ID, but when you answer, there's no one there because it's not a real call.
* To stop them, look for a setting in your phone such as:
**SIP Trust
**Restrict Source IP
**Direct IP calls
**Accept SIP Trust Server Only eg, [[VoIP_Phones_-_Yealink_DECT#Other_optional_settings|Yealink Phones have this option]]
**Allow Incoming SIP Messages from SIP Proxy Only [[VoIP_Phones_-_Grandstream_HT8xx#Ghost_calls?|Grandstream Phones have this option]]
* Also review your [[VoIP Security|Firewall]], and check you're only allowing VoIP traffic from the A&A servers.
==Audio/Sound problems==
* '''Also See: [[SIP Audio Problems]] We have an extensive checklist to help diagnosing audio quality problems'''
▲===One-way audio===
One way audio is usually due to network issues where NAT is in use.
*Check if your router has a 'SIP ALG' enabled. Disable this if so. See: [[Disable SIP ALG]]
===Calls breaking up (intermittent audio)===
Generally audio problems such as calls breaking up are due to to the quality of the Internet connected that is being used. Diagnostics should first be concentrated on how well the internet connection is working.
*Try removing everything from your broadband router and directly plug in a single phone in to the router - this will eliminate the local network
*See: [[SIP Audio Problems]] We have an extensive checklist to help diagnosing audio quality problems
=== One-way audio on call recordings ===
Our call recording system will produce audio with each call participant on separate stereo audio channels. Make sure you are listening with both earbuds in. If you are finding that you are still only hearing one person, then contact our Support team supplying the recording in question.
=== Sped up or slowed down call recordings ===
This can happen when you encounter a codec issue on the machine you are listening with. We would suggest using WAV or MP3 for call recordings. This can be adjusted on the [http://control.aa.net.uk Control Panel].
[[Category:Faults]]
[[Category:VoIP]]
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