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Re: [LUG] VoIP and ISP

 

Henry Bremridge wrote:
> On Wed, Dec 24, 2008 at 05:29:30PM +0000, Mark Evans wrote:
>   
>> What audio codec is being used? If it's either GSM or G.711 (A-law)
>> there should be no problems
>> because there will either be no translation required or translation
>> should only take place at
>> PSTN interconnect. Check you arn't using G.711 (Mu-law)...
>>
>>     
>
> Phone does not give the information on codecs. In the control panel there is a 
> bandwidth saver option: the allows the user to choose which of the following 
> settings were used:
> - 30 kpbs
> - 50 kpbs
> - 90 kpbs
>
> The phone is set at the highest setting, but following Gordons comments have moved 
> the vonage from DECT to landline. Am waiting to hear if there has been any 
> improvement
>
> Currently ploughing through the asterisk book... :(
>
>
>   
VoIP should be usable on any of the current DSL providers.. I have 
Asterisk running on a Dell D600 laptop (its broken screen disconnected)  
this is slotted into a rack and has  been in constant use for  over 3 
years.  I  use several different  VoIP providers inc Sipgate,Voiptalk, 
Heart Telecom and even Tesco Internetphone  and apart from tesco not 
registering when dynamic IP changes - a quick reboot sorts this out - I 
have had no problems at all - voice quality is comparable with standard 
landline.

I use SIP internally with both hard & softphones and IAX2 and SIP (with 
STUN) on the VoIP trunks. The PSTN is handled via Sipura SPA3000 and I 
have several hard phones connected throughout the house and office, a 
couple of SIP hard phones with the rest being normal phones connected to 
ATA's - no problems at all

I'm currently messing about with a GSM gateway connected to another 
Asterisk box via an internal X100p card - I have it working - just need 
to connect this box to the Dell.

I will say this tho' - all my VoIP traffic is on its own network having 
its own wiring, switches etc and only 'meets' the outside world at the 
firewall -  no LAN traffic!!



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