Hello Gareth,
echo also appears when making calls with a SIP phone. These are outgoing
calls.
Another site now also gives feedback on echo, telling they sometimes
also have echo on outgoing calls and if they recall right then sometimes
also on incoming calls (coming from a queue).
This
Hello list,
this is the setup :
analogue phone -- Grandstream GXW4008 -- Linksys WAG160N --
Asterisk-server (public)
and
Zoiper softphone -- Linksys WAG160N -- Asterisk-server (public)
When calling with an analogue phone + Grandstream GXW and also when
calling with the Zoiper softphone, we
Jonas Kellens wrote:
Hello list,
this is the setup :
analogue phone -- Grandstream GXW4008 -- Linksys WAG160N --
Asterisk-server (public)
and
Zoiper softphone -- Linksys WAG160N -- Asterisk-server (public)
When calling with an analogue phone + Grandstream GXW and also when
Hello,
I also thought about echo because the Zoiper softphone is used with a
headset. But that didn't explain why the echo also appeared on the
analogue phone + gateway.
I have the same Grandstream GXW 4008 gateway with 5 analoge phones
attached in another environment and there, there are
Routers wont cause echo. In order for them to do so they would have to
store the outbound voice traffic, delay it and then mix it into the
inbound voice.
Telephones inherently cause echo. For domestic calls the audio path is
normally so short that any echo arrives back so quick the human ear
On 30 June 2010 10:28, Jonas Kellens jonas.kell...@telenet.be wrote:
Hello,
I also thought about echo because the Zoiper softphone is used with a
headset. But that didn't explain why the echo also appeared on the analogue
phone + gateway.
It will present it self on the analogue phone when
Hello,
I stated in my first post that both ends hear an echo when one speaks to
the other...
The only place where echo cancellation is being applied is in the
Asterisk server. I have the following in sip.conf :
;-- JITTER BUFFER CONFIGURATION
Thats the jitter buffer. It has no effect on echo.
So you get echo when calling from the softphone to the analogue phone?
What about when one of those calls somewhere else?
What if they call a regular telephone number?
How do you connect in order to send calls to normal phone numbers?
Jonas
Hello,
I did not say that the analogue phone calls the Zoiper softphone or vica
versa.
Calls are made to from the Zoiper to an external number like a cellphone.
Calls are also made from the analogue phone to external numbers like an
international number in Holland...
Jonas.
On 30 June
On 06/30/2010 12:20 PM, Gareth Blades wrote:
So you get echo when calling from the softphone to the analogue phone?
From softphone to analogue phone is echo.
What if they call a regular telephone number?
Calling to a cellphone number or a fixed number on another Telco-network
: echo
Jonas Kellens wrote:
On 06/30/2010 12:20 PM, Gareth Blades wrote:
So you get echo when calling from the softphone to the analogue phone?
From softphone to analogue phone is echo.
What if they call a regular telephone number?
Calling to a cellphone number or a fixed number on
Internet Telephony Service Provider = SIP provider. The company that
connects the Asterisk-server via a SIP trunk with the other networks
like GSM, analogue carriers...
Jonas.
By ITSP do you mean a SIP provider?
--
_
On 30 Jun 2010, at 13:48, Gareth Blades wrote:
By ITSP do you mean a SIP provider?
ITSP: Internet Telephony Service Provider
S
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join
Hi!
The network setup is :
analogue+GXW / softphone -- Linksys WAG160N -- Asterisk server -- ITSP
-- other networks
Do it step-by-step: Take the Asterisk server out of the equation, i.e.
call the destination directly with your softphone or the Grandstream ATA
and see if that removes the
Jonas Kellens wrote:
Internet Telephony Service Provider = SIP provider. The company that
connects the Asterisk-server via a SIP trunk with the other networks
like GSM, analogue carriers...
Jonas.
By ITSP do you mean a SIP provider?
Thats where I believe the problem lies. You are
Gareth,
multiple users/SIP-accounts use this asterisk server from many
locations. Like I said: in another location with a similar setup, there
are no echo-complaints on received or made calls.
If you say that it has nothing to do with the Cisco-router, I don't
really know what to go looking
Try the SIP phone. If it is better then you might try looking to see if
there are any echo cancelation settings on the softphone or analogue
adapter you can change. Try turning echo cancelation off aswell since if
there are two running they can interfere with each other and make the
situation
Will turning off the jitter buffer affect the quality of the other calls ??
jbenable = no
I must say I'm not really into these jitter-settings in asterisk. I made
jbenable=yes as it can do no harm...
Jonas.
On 06/30/2010 04:24 PM, Gareth Blades wrote:
Try the SIP phone. If it is better
-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Wednesday, June 30, 2010 9:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Echo problem in VoIP-calls
Will turning off the jitter
Yes if you have a link where there is a lot of jitter it may affect the
call quality. I would try turning it off to see if it cures the problem
and if it does then you can restore the setting and implement a workaround.
Jonas Kellens wrote:
Will turning off the jitter buffer affect the quality
20 matches
Mail list logo
