First of all, Alex, sorry for not seeing your reply. Nearly two weeks
ago now :(
Honestly, with canreinvite=yes, I'm not sure what is meant by the
signalling still travels through asterisk... I would ASSUME that
includes out-of-band dtmf as well. Sorry!
Moj
Alex Crow wrote:
Moj,
Does
:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Deepak
Naidu
Sent: Tuesday, June 12, 2007 19:28
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Bad Echo between SIP calls
I like the way people replied to this message of mine. It seems
On Jun 11, 2007, at 7:26 PM, Zeeshan Zakaria wrote:
Also I recommend going with Sangoma. I hear a lot of bad stories about
digium cards imcompatibility with certain motherboards and conflicts
with USB modules on the motherboard, and conflicts with IRQs. Thats
why When I went for PRI, I used
theoretically, with canreinvite=yes, it's phone - phone. with
canreinvite=no, it's phone - asterisk - phone. BUT there are a few
reasons which canreinvite=yes will not be this way. If for example you
have a T or a t in the Dial string, asterisk will _remain_ in the media
path so it can
Moj,
Does this mean that even out-of-band DTMF still gets sent
SIP-phone--SIP-phone without Asterisk hearing them? (eg RFC DTMF,
can't remember the number right now)
Forgive me for butting into this thread but this is interesting...
Cheers
Alex
On Tue, 2007-06-12 at 09:21 -0800, Mojo
I don't see this listed anywhere here in the replies so.
In your zapata.conf file try changing:
echocancelwhenbridged=no
to:
echocancelwhenbridged=yes
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
Remember to restart asterisk and zaptel when you make this change.
On 6/12/07, Matt [EMAIL PROTECTED] wrote:
I don't see this listed anywhere here in the replies so.
In your zapata.conf file try changing:
echocancelwhenbridged=no
to:
echocancelwhenbridged=yes
On Tue, Jun 12, 2007 at 07:44:02PM -0400, Matt wrote:
On 6/12/07, Matt [EMAIL PROTECTED] wrote:
I don't see this listed anywhere here in the replies so.
In your zapata.conf file try changing:
echocancelwhenbridged=no
to:
echocancelwhenbridged=yes
Remember to restart asterisk and
: [asterisk-users] Bad Echo between SIP calls
I don't see this listed anywhere here in the replies so.
In your zapata.conf file try changing:
echocancelwhenbridged=no
to:
echocancelwhenbridged=yes
___
--Bandwidth and Colocation provided by Easynews.com
.
-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Tuesday, June 12, 2007 16:44
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bad Echo between SIP calls
I don't see this listed anywhere here in the replies so
: [asterisk-users] Bad Echo between SIP calls
I like the way people replied to this message of mine. It seems this
thread is going back to the hybrid echo issue(no this is not the
problem). As said by many ZAP is not in picture for SIP--SIP ie
Ext-Ext internal calls.
To put my inputs I did tons
Deepak Naidu wrote:
I like the way people replied to this message of mine. It seems this thread is
going back to the hybrid echo issue(no this is not the problem). As said by
many ZAP is not in picture for SIP--SIP ie Ext-Ext internal calls.
To put my inputs I did tons of QA on this
On Jun 8, 2007, at 6:00 PM, Deepak Naidu wrote:
Hi,
We have a PRI connection when its was on test networks we
had echo problems withoutside line.
So I bought a TE212P card resolve the echo problem. Which did to an
extent. Its using asterisk 1.2.18 RHEL4-Update 4.
But now
Sounds crazy right? even was I, more over support guy logged in unloaded the
zap modules to test them, still an echo.
Ya, I was clear saying that we have SIP--- SIP issue ie internal extension echo
problem. It seems the echo with SIP--SIP has many factors. I am just curios
to eliminate any
are using.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Deepak
Naidu
Sent: Monday, June 11, 2007 16:37
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bad Echo between SIP calls
Sounds crazy right? even
Once upon a time I used to have a lot of SIP-SIP calls issues, which not
always but sometimes included echo problems. There were no zap devices on
the server. Googling and struggling to fix it, I found out that it was
because of timing issues and ztdummy was not working properly. It had to do
Hey thanx for sharing your troubleshooting. Ya over days I kind of did some
QA. There are SIP--SIP echo's between random phones. We have 75 phones of
Polycom 501. I think might be the network or combination of network polycom
creating this.
Do you have the backup of old setup without
PROTECTED] On Behalf Of C F
Sent: Saturday, June 09, 2007 22:54
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bad Echo between SIP calls
It doesn't matter if it's supported, they are all, however I have seen
some echo problems after firmware upgrades
Deepak Naidu wrote:
Steve I understand your theory. We have Poycom 501 phones. Prior upgrading to PRI
we were till date using 4 analog lines connected with TDM card from digium no
echo for pure SIP to SIP lines.
Now I have TE212P which had onboard echo cancellor.
I am trying
On 09/06/07, Deepak Naidu [EMAIL PROTECTED] wrote:
Ya, I have done that, below is zapata.conf. Also we had an TMP card with
analog lines. SIP cals were great on them. now when we switched over.
SIP calls have echo.. which shouldnt be at all.
If you are getting echo on pure SIP to SIP calls,
Steve I understand your theory. We have Poycom 501 phones. Prior upgrading to
PRI we were till date using 4 analog lines connected with TDM card from digium
no echo for pure SIP to SIP lines.
Now I have TE212P which had onboard echo cancellor.
I am trying make myself clear before
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stephen Davies
Sent: Saturday, June 09, 2007 4:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bad Echo between SIP calls
On 09/06/07
:[EMAIL PROTECTED] On Behalf Of Deepak
Naidu
Sent: Saturday, June 09, 2007 4:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bad Echo between SIP calls
Steve I understand your theory. We have Poycom 501 phones. Prior
upgrading to PRI we were till date
Yeah I have made sure its the correct port. We have 75 polycoms currently.
? the SIP-to-SIP echo is there.
--
Deepak
Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote:
Deepak Naidu wrote:
Steve I understand your theory. We have Poycom 501 phones. Prior upgrading to
PRI we were till
Subject: Re: [asterisk-users] Bad Echo between SIP calls
Steve I understand your theory. We have Poycom 501 phones. Prior
upgrading to PRI we were till date using 4 analog lines connected with TDM card
from digium no echo for pure SIP to SIP lines.
Now I have TE212P which
Stephen Davies wrote:
On 09/06/07, Deepak Naidu [EMAIL PROTECTED] wrote:
Ya, I have done that, below is zapata.conf. Also we had an TMP card
with
analog lines. SIP cals were great on them. now when we switched
over.
SIP calls have echo.. which shouldnt be at all.
If you are getting echo
Are the config files you are using with the phones what was meant with
that firmware? or did you upgrade the firmware and reused the old
config files?
On 6/9/07, Steve Underwood [EMAIL PROTECTED] wrote:
Stephen Davies wrote:
On 09/06/07, Deepak Naidu [EMAIL PROTECTED] wrote:
Ya, I have done
The sip config firmware are the supported one for the existing firmware. If
you have any stable working Polycom 501 SIP without echo between SIP--SIP
wouldnt mind to share the sip.cfg, sip.ld bootrom would be great, bcos I have
not got concreate resolution for this issue.
Hope I can
It doesn't matter if it's supported, they are all, however I have seen some
echo problems after firmware upgrades, the only way to fix it was to either
copy the differences or overwrite my old config files with the new ones that
came with the firmware and then modify as needed for my setup.
On
Hi,
We have a PRI connection when its was on test networks we had echo
problems withoutside line.
So I bought a TE212P card resolve the echo problem. Which did to an extent.
Its using asterisk 1.2.18 RHEL4-Update 4.
But now when we are live, there is a terrible echo between 2
On Sat, 9 Jun 2007, Deepak Naidu wrote:
But now when we are live, there is a terrible echo between 2 SIP calls.
If I call the same extension from outside the voice is clear.
My impression is that the transcoding that takes place between two
purely software SIP calls never goes through the
Ya, I have done that, below is zapata.conf. Also we had an TMP card with
analog lines. SIP cals were great on them. now when we switched over. SIP
calls have echo.. which shouldnt be at all.
[channels]
language=en
#include zapata_additional.conf
context=from-pstn
switchtype=national
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