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uk.telecom.voip (UK VOIP) (uk.telecom.voip) Discussion of topics relevant to packet based voice technologies including Voice over IP (VoIP), Fax over IP (FoIP), Voice over Frame Relay (VoFR), Voice over Broadband (VoB) and Voice on the Net (VoN) as well as service providers, hardware and software for use with these technologies. Advertising is not allowed.

SipGate & Echo - Need alternative?



 
 
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  #1  
Old June 19th 07, 12:19 PM posted to uk.telecom.voip
elziko
external usenet poster
 
Posts: 9
Default SipGate & Echo - Need alternative?

I have been using SipGate or months but have suffered a terrible echo that
the other caller seems to hear. The call from my end sounds fine.

With these problems SipGate just demand that I fill out a form for every
phone call and then write back to me to tell me that their servers were
running fine at the time the calls were made.

Can anyone suggest a reason for the person calling hearing a bad echo whilst
everything sounds fine for me?

I am using an SPA-2000 with ports 560 & 16384 forwarded. As requested by
SipGate support I am not using the STUN server. I can let you know about any
other seetings that may be relevant.

Are there any alternative providers that offer no monthly fee and a free
geographical number?

TIA


  #2  
Old June 19th 07, 12:34 PM posted to uk.telecom.voip
Tim
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Posts: 385
Default SipGate & Echo - Need alternative?

elziko wrote:
I have been using SipGate or months but have suffered a terrible echo that
the other caller seems to hear. The call from my end sounds fine.


This isn't going to be SipGate's problem.

Can anyone suggest a reason for the person calling hearing a bad echo whilst
everything sounds fine for me?


Yes. Audio is leaking from your ear piece into your microphone.

Have you tried plugging a different phone into the SPA-2000?

Do you have the correct impedance value set on the SPA-2000? To match
the phone you using.

Have you tried turning down the microphone gain on your end?


I am using an SPA-2000 with ports 560 & 16384 forwarded. As requested by
SipGate support I am not using the STUN server. I can let you know about any
other seetings that may be relevant.


Port forwarding and stun have nothing at all to do with audio quality.

Tim
  #3  
Old June 19th 07, 02:32 PM posted to uk.telecom.voip
elziko
external usenet poster
 
Posts: 9
Default SipGate & Echo - Need alternative?

Thanks for the reply!

Tim wrote:
Have you tried plugging a different phone into the SPA-2000?


Not yet, I'll find a replacement and give it a go. But in the meantime...

Do you have the correct impedance value set on the SPA-2000? To match
the phone you using.


Can you give me any information on where on the SPA-2000 I'd adjust this? I
assume it'll be in the 'Line 1' settings since all other seetings will
affect both 'Line 1' and 'Line 2' where the latter could have a totally
different phone on it. Also how would I go about choosing a value for this?

Have you tried turning down the microphone gain on your end?


If you mean the gain on my actual telephone - I don;t believe I can set
this. If there is an additional gain built into the ATA then, as above, can
you tell me where I'd find it? I'd expect it to be under 'Line 1' again but
don;t seem to be able to see anything that looks relevant.

TIA


  #4  
Old June 19th 07, 02:39 PM posted to uk.telecom.voip
Graham
external usenet poster
 
Posts: 188
Default SipGate & Echo - Need alternative?


"elziko" wrote in message
...
I have been using SipGate or months but have suffered a terrible echo that
the other caller seems to hear. The call from my end sounds fine.

With these problems SipGate just demand that I fill out a form for every
phone call and then write back to me to tell me that their servers were
running fine at the time the calls were made.

Can anyone suggest a reason for the person calling hearing a bad echo
whilst everything sounds fine for me?

I am using an SPA-2000 with ports 560 & 16384 forwarded. As requested by
SipGate support I am not using the STUN server. I can let you know about
any other seetings that may be relevant.

Are there any alternative providers that offer no monthly fee and a free
geographical number?

TIA


You don't tell us what you have got plugged into the phone-port of
the ATA. Is it DECT or a standard wired phone. Do you use it
as a hands free speakerphone?

--
Graham.
%Profound_observation%


  #5  
Old June 19th 07, 02:44 PM posted to uk.telecom.voip
elziko
external usenet poster
 
Posts: 9
Default SipGate & Echo - Need alternative?

Graham wrote:
You don't tell us what you have got plugged into the phone-port of
the ATA. Is it DECT or a standard wired phone. Do you use it
as a hands free speakerphone?


Sorry, it is a DECT phone but i do not use it as a speaker phone.


  #6  
Old June 19th 07, 02:59 PM posted to uk.telecom.voip
Graham
external usenet poster
 
Posts: 188
Default SipGate & Echo - Need alternative?


"elziko" wrote in message
...
Thanks for the reply!

Tim wrote:
Have you tried plugging a different phone into the SPA-2000?


Not yet, I'll find a replacement and give it a go. But in the meantime...

Do you have the correct impedance value set on the SPA-2000? To match
the phone you using.


Can you give me any information on where on the SPA-2000 I'd adjust this?
I assume it'll be in the 'Line 1' settings since all other seetings will
affect both 'Line 1' and 'Line 2' where the latter could have a totally
different phone on it. Also how would I go about choosing a value for
this?

Have you tried turning down the microphone gain on your end?


If you mean the gain on my actual telephone - I don;t believe I can set
this. If there is an additional gain built into the ATA then, as above,
can you tell me where I'd find it? I'd expect it to be under 'Line 1'
again but don;t seem to be able to see anything that looks relevant.

TIA


My device is a Linksys PAP-2 so it might be a bit different.
Look in the Regional tab under miscellaneous
FSX Port impedence = 370+620||310nF

Play around with the FSX input and output gain
mine are both set to -3 (minus 3)

In the Line 1 and Line 2 tabs, audio configuration make
sure all the options related to echo are on (there are three on the PAP2)

Apologies if any of this is not relevant to your device.


--
Graham.
%Profound_observation%


  #7  
Old June 19th 07, 03:18 PM posted to uk.telecom.voip
elziko
external usenet poster
 
Posts: 9
Default SipGate & Echo - Need alternative?

Graham wrote:
My device is a Linksys PAP-2 so it might be a bit different.
Look in the Regional tab under miscellaneous


Thanks i found them - the settings are the same as yours but i will have a
play to see if I can improve things.


  #8  
Old June 19th 07, 03:58 PM posted to uk.telecom.voip
{{{{{Welcome}}}}}
external usenet poster
 
Posts: 908
Default SipGate & Echo - Need alternative?

elziko wrote:
Graham wrote:
My device is a Linksys PAP-2 so it might be a bit different.
Look in the Regional tab under miscellaneous


Thanks i found them - the settings are the same as yours but i will
have a play to see if I can improve things.


The FXS Port Impedance needs to be set to: 370+620||310nF if using
phones designed for use in the UK

As mentioned above drop the drop the gain slightly and check the various
echo reduction options, you'll have to go into admin and advanced mode
to see some of these.

Other settings you may find and may want to check just to get a UK feel
to the device if not already done so (these are bassed on a PAP2, so may
be slight difference on which are listed):


Under Regional Tab.

Dial Tone: [email protected],[email protected];60(*/0/1+2)
Second Dial Tone: [email protected],[email protected];10(*/0/1+2)
Outside Dial Tone: [email protected];10(*/0/1)
Prompt Tone: [email protected],[email protected];10(*/0/1+2)
Busy Tone: /1)
Reorder Tone: [email protected];20(*/0/1)
Off Hook Warning Tone: /1+2)
Ring Back Tone: /2/1+2)
Confirm Tone: /1)
SIT1 Tone:
/0/3,0/1/0)
SIT2 Tone:
/0/3,0/4/0)
SIT3 Tone:
/0/3,0/4/0)
SIT4 Tone:
/0/3,0/4/0)
MWI Dial Tone: /1+2)
Cfwd Dial Tone: /1+2);10(*/0/1+2)
DND Dial Tone: /2);10(*/0/1+2)
Holding Tone: /1)
Conference Tone: /1)
Secure Call Indication Tone: /2)
Feature Invocation Tone: /1)

Ring1 Cadence: 60(.4/.2,.4/2)
Ring2 Cadence: 60(1/2)
Ring3 Cadence: 60(.25/.25,.25/.25,.25/1.75)

....

CWT1 Cadence: 30(.2/.2,.2/4.4)

Ring Waveform: Sinusoid (though Trapezoid may help problematic phones to
ring).
Ring Frequency: 25
Ring Voltage: 80 (You can use 70, if problems try 75, 80, 85, 90)
CWT Frequency: [email protected]
Synchronized Ring: Yes

Hook Flash Timer Min: .06
Hook Flash Timer Max: .2
Callee On Hook Delay: 0
Reorder Delay: 5
Call Back Expires: 1800
Call Back Retry Intvl: 30
Call Back Delay: .5
VMWI Refresh Intvl: 0
Interdigit Long Timer: 10
Interdigit Short Timer: 3
CPC Delay: .5
CPC Duration: .1

....

Time Zone: GMT
FXS Port Impedance: 370+620||310nF
Daylight Saving Time Rule:
start=3/-1/7/2:0:0;end=10/-1/7/2:0:0;save=1:0:0
DTMF Playback Level: -16
DTMF Playback Length: .25
Caller ID Method: ETSI FSK With PR(UK)
FXS Port Power Limit: 3
Caller ID FSK Standard: v.23
Feature Invocation Method: Default
More Echo Suppression: yes

In Line 1 (or Line 2)

Line Enable: yes
NAT Mapping Enable: yes
NAT Keep Alive Enable: yes
Network Jitter Level: low (Depends on how good your broadband is / route
to VoIP server)
Jitter Buffer Adjustment: Up and Down


Proxy and Registration
Proxy: Your Voip Proxy address (for example sip1.sipdiscount.com or
sip.voipcheap.com or sipgate.co.uk etc)

Subscriber Information - (where you enter your VoIP Account details).

Supplementary Service Subscription - This is where you enable services
such as Call Waiting, 3-Way Calling, block anonymous caller etc)

Audio Configuration - This is where you set-up which codecs you want to
use or only use.
If you have limited broadband speed (mostly if upload is poor) then you
probably only use one or more from G729a, G723 or G726)
If you want the best quality line, then you'd choose G711a (which is the
standard used on a standard line)

(Myself, I use G711a only, have that set to my preferred codec and use
that one only, but you may find other codecs are fine).

DTMF Tx Method: (This will depend on the VoIP service you use, trial and
error and depends on codec) Soon know if it don't work when
try telephone banking for example and it
doesn't recognise what numbers you are pressing).

Hook Flash Tx Method: None

 




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