Category:VoIP Faults: Difference between revisions
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With most VoIP problems it is best to contact Support. However, here are a few pointers: |
With most VoIP problems it is best to contact Support. However, here are a few pointers: |
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== Problems connecting to our servers == |
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[[File:Snom-notregistered.png|right|frame|The screen of a Snom]] |
[[File:Snom-notregistered.png|right|frame|The screen of a Snom]] |
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*Check username and password carefully. (Typically the username is the full phone number in international format, eg +44.....) |
*Check username and password carefully. (Typically the username is the full phone number in international format, eg +44.....) |
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*Check if your router has a SIP ALG - see: [[Disable_SIP_ALG]] |
*Check if your router has a SIP ALG - see: [[Disable_SIP_ALG]] |
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== Missing calls, or strange calls == |
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=== Only some calls are getting through === |
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=== Using an ATA but the phone doesn't actually ring, but it still works === |
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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/ |
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=== Ghost calls /phantom calls / Strange CLI, nobody there=== |
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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. |
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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. |
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* To stop them, look for a setting in your phone such as: |
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**SIP Trust |
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**Restrict Source IP |
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**Direct IP calls |
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**Accept SIP Trust Server Only eg, [[VoIP_Phones_-_Yealink_DECT#Other_optional_settings|Yealink Phones have this option]] |
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**Allow Incoming SIP Messages from SIP Proxy Only [[VoIP_Phones_-_Grandstream_HT8xx#Ghost_calls?|Grandstream Phones have this option]] |
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* Also review your [[VoIP Security|Firewall]], and check you're only allowing VoIP traffic from the A&A servers. |
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==Audio/Sound problems== |
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* '''Also See: [[SIP Audio Problems]] We have an extensive checklist to help diagnosing audio quality problems''' |
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One way audio is usually due to network issues where NAT is in use. |
One way audio is usually due to network issues where NAT is in use. |
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*Check if your router has a 'SIP ALG' enabled. Disable this if so. See: [[Disable SIP ALG]] |
*Check if your router has a 'SIP ALG' enabled. Disable this if so. See: [[Disable SIP ALG]] |
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==Calls breaking up (intermittent audio)== |
===Calls breaking up (intermittent audio)=== |
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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. |
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. |
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*Try removing everything from your broadband router and directly plug in a single phone in to the router - this will eliminate the local network |
*Try removing everything from your broadband router and directly plug in a single phone in to the router - this will eliminate the local network |
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*See: [[ |
*See: [[SIP Audio Problems]] We have an extensive checklist to help diagnosing audio quality problems |
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=== One-way audio on call recordings === |
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== One-way audio on call recordings == |
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[[Category:Faults]] |
[[Category:Faults]] |