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Standard User 477
(newbie) Sat 12-Oct-13 11:34:18
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Gigaset N300A IP Voip Phone


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Hi

I have a Gigaset N300A IP Voip phone base station and 2 Gigaset SL78H handsets. When I make a call the connection is made successfully but it drops out part way through the call and if I try to redial the line is busy. There is no set time that the drop-out occurs it is just random but it always happens. I have changed the base station for a new one, checked the compatibility/suitability of my router and checked the satellite broadband latency and up/download speeds and all appears to be ok. Has anyone had similar problems and resolved them successfully or can anyone help me to resolve them?
Standard User David_W
(fountain of knowledge) Sat 12-Oct-13 16:55:14
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Re: Gigaset N300A IP Voip Phone


[re: 477] [link to this post]
 
In reply to a post by 477:
checked the satellite broadband latency
If you are really on satellite, that could be your issue. VoIP is sensitive to packet loss and latency.


However, the issue may be with your router, especially if it's working with a single IP address using Network Address Translation (NAT). There are various ways to deal with SIP behind a NAT router, with some of these requiring assistance from the router via something called an Application Layer Gateway (ALG).

The issue may also be the SIP connection timing out and being closed by the router's firewall.

Standard User ferretuk
(regular) Sat 12-Oct-13 18:18:49
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Re: Gigaset N300A IP Voip Phone


[re: 477] [link to this post]
 
Have you tried a software phone? Sounds to me that this is connection related rather than anything to do with your hardware.


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Standard User Mr_Morpheus
(newbie) Sat 12-Apr-14 11:12:46
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Re: Gigaset N300A IP Voip Phone


[re: ferretuk] [link to this post]
 
Not necessarily, if the issue is being caused by NAT timers closing ports, or the WAN connection changing the client won't make much difference.

The advantage of using a PC/MAC based client is that you can view the SIP logs and if necessary run Wireshark to capture the low-level traffic.

Don't suppose you can clarify the router you are using? Most VoIP providers require that the SIP ALG component of the router/firewall is disabled as their services generally handle NAT translation themselves and having the ALG on does often cause issues (usually the symptom of this is the 30/32 second disconnect as the final ACK or 200 OK gets missed)...

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I am NOT affiliated with ANY ISP on thinkbroadband.com, I have used various ISPs and I AM knowledgeable on the subject of broadband.

"Seek and ye shall find"
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