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 SIP calls. If I
call the same extension from outside the voice is clear.
I am not sure whats the problem. Also there's slight echo when calling
Digium support.
Totally lost Digium says we need to remove the echo module to resolve SIP echo
problems. Then ? the heck we pay for...
Has anyone come through this issue.
--
Deepak
---------------------------------
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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 TE212P card. There are probably echo cancellation options you can enable that are relevant to software channels. I distantly recall there even being some stuff youc an uncomment in the source.> Totally lost Digium says we need to remove the echo module to resolve > SIP echo problems. Then ? the heck we pay for...Not sure why Digium would say that. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671
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 pridialplan=national signalling=pri_cpe faxdetect=incoming usecallerid=yes echocancel=yes callerid=asreceived echocancelwhenbridged=no echotraining=128 ;rxgain=-3.0 ;txgain=-7.0 group=0 channel=1-23 -- Deepak Alex Balashov <abalashov@evaristesys.com> wrote: 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 TE212P card. There are probably echo cancellation options you can enable that are relevant to software channels. I distantly recall there even being some stuff youc an uncomment in the source.> Totally lost Digium says we need to remove the echo module to resolve > SIP echo problems. Then ? the heck we pay for...Not sure why Digium would say that. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671 _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --------------------------------- All New Yahoo! Mail ? Tired of unwanted email come-ons? Let our SpamGuard protect you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070608/b70e7d58/attachment.htm
> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] 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, Deepak Naidu <deepak_nai@yahoo.com> 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, there's no point in > fiddling around with your zapta.conf. That file is for configuring > chan_zap, which is used to talk to Zap/ channels. Your calls are SIP > to SIP so the zap channel and your PRI aren't being used at all. > > SIP calls are "pure digital" 4 wire lines so no electrical (Hybrid) > echo will be present. The phones should not generate echo. If they > are, they are presumably nasty phones (what kind are they?) and you > should get properly made phones. > > SteveMost likely the phones. Is it worse on speakerphone? Are they cheap like the Grandstream 101s? Try with a couple softphones and headsets, any better. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB
Do you have reinvites enabled? Are you running this over a linksys four port SoHo router/switch or something? Thanks, Steve Totaro http://www.asteriskhelpdesk.com <http://www.asteriskhelpdesk.com/component/option,com_wrapper/Itemid,37/>KB3OPB _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] 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 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 I blame on any network. B'cos for sure we have a spegati of networks & no QoS. Also the intresting thing is if I call from one extension to other dialing the main line & then extension the call is crystal clear. but when dialing a direct extension its a hell of echo. -- Deepak Stephen Davies <stephen.l.davies@gmail.com> wrote: On 09/06/07, Deepak Naidu 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, there's no point in fiddling around with your zapta.conf. That file is for configuring chan_zap, which is used to talk to Zap/ channels. Your calls are SIP to SIP so the zap channel and your PRI aren't being used at all. SIP calls are "pure digital" 4 wire lines so no electrical (Hybrid) echo will be present. The phones should not generate echo. If they are, they are presumably nasty phones (what kind are they?) and you should get properly made phones. Steve _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _____ Yahoo! Answers - Get better answers from someone who knows. Try it now <http://uk.answers.yahoo.com/;_ylc=X3oDMTEydmViNG02BF9TAzIxMTQ3MTcxOTAEc 2VjA21haWwEc2xrA3RhZ2xpbmU> . -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070609/b0534a52/attachment.htm
Ibrar Ahmed
2007-Jun-09 05:03 UTC
[asterisk-users] Is There any Asterisk TODO(Developer side) list Available
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Best way to do this is not touch the sip.cfg, ever. Leave it as included
in each release and include your overrides in a different file.
Then reference your files like this in your MAC.cfg file, your file will
override the sip.cfg defaults.
CONFIG_FILES="phone_user.cfg,server.cfg,sip.cfg"
In server.cfg, if you wanted to change the server, for example:
<?xml version="1.0" standalone="yes"?>
<sip>
<voIpProt>
<local voIpProt.local.port=""/>
<server voIpProt.server.1.address="asterisk.yourdomain.com"
</voIpProt>
</sip>
________________________________
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] 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, 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 6/10/07, Deepak Naidu <deepak_nai@yahoo.com> wrote:
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 resolve this mess. Feels bad when one does best in
aggregating things & some louzy device screws up... Oh my frustation is
comming on mail :
<http://us.i1.yimg.com/us.yimg.com/i/mesg/tsmileys2/03.gif>
--
Deepak
C F < shmaltz@gmail.com <mailto:shmaltz@gmail.com> > wrote:
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 wrote:
> Stephen Davies wrote:
> > On 09/06/07, Deepak Naidu 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,
there's no point in
> > fiddling around with your zapta.conf. That file is
for configuring
> > chan_zap, which is used to talk to Zap/ channels.
Your calls are SIP
> > to SIP so the zap channel and your PRI aren't being
used at all.
> >
> > SIP calls are "pure digital" 4 wire lines so no
electrical (Hybrid)
> > echo will be present. The phones should not generate
echo. If they
> > are, they are presumably nasty phones (what kind are
they?) and you
> > should get properly made phones.
> By this measure most phones are nasty. The handset
should be echo
> cancelled, to prevent leakage of the earpiece into the
mike. It is
> getting less and less common to do this, now.
Polycoms, Sipuras, Snoms,
> you name it, they do it badly. Many are not too
annoying until someone
> turns the volume up. Call someone a little hard of
hearing and you will
> hear echo.
>
> Steve
>
>
> _______________________________________________
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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 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. > > I am not sure whats the problem.? Also there's slight echo when > calling Digium support. > > Totally lost Digium says we need to remove the echo module to resolve > SIP echo problems. Then ? the heck we pay for..Are you sure that they understood that you were having this problem between 2 SIP endpoints? That advice only makes sense to test if one side is Zap and the other side is SIP. --- Matthew Fredrickson Software Engineer Digium, Inc.
The echo cancellation card is for SIP->Zap calls only, no echo cancellation is done in Asterisk for SIP only calls. SIP to SIP, media is just passed through the server untouched (using media flow through, which is the option in sip.conf of canreinvite=no) if you are not handling any translation, even when handling translation between SIP calls there shouldn't be any echo cancellation done in Asterisk for SIP only calls. The place to look at would be the remote SIP devices which is typically what is adding the echo, this is usually a gain issue of some sort depending on which handsets you are using. ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] 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 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 possibility of Asterisk failing to cancel the echo. OK, one question here howz the call flow when a SIP---SIP call is established ie. is the connection between 2 phones when an Internal call is made or does the SIP call goes via Asterisk once the SIP--SIP call is establised. -- Deepak Matthew Fredrickson <creslin@digium.com> wrote: 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 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. > > I am not sure whats the problem. Also there's slight echo when > calling Digium support. > > Totally lost Digium says we need to remove the echo module to resolve > SIP echo problems. Then ? the heck we pay for.. Are you sure that they understood that you were having this problem between 2 SIP endpoints? That advice only makes sense to test if one side is Zap and the other side is SIP. --- Matthew Fredrickson Software Engineer Digium, Inc. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ________________________________ Yahoo! Answers - Get better answers from someone who knows. Try it now <http://uk.answers.yahoo.com/;_ylc=X3oDMTEydmViNG02BF9TAzIxMTQ3MTcxOTAEc 2VjA21haWwEc2xrA3RhZ2xpbmU> . -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070611/755d52d0/attachment.htm
I don't see this listed anywhere here in the replies so. In your zapata.conf file try changing: echocancelwhenbridged=no to: echocancelwhenbridged=yes -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070612/9d3aeea9/attachment-0001.htm
This should only be for TDM to TDM calls, SIP to SIP calls don't use the zaptel driver. ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] 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. In your zapata.conf file try changing: echocancelwhenbridged=no to: echocancelwhenbridged=yes -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070612/9d55285d/attachment.htm
What are the end devices? That seems to have been lost here. The real issue is the handsets as those are the devices introducing the echo (the only analog players here). Most likely a volume or gain issue on those handsets, what SIP devices are the echo issues between? If both people hear echo, both devices are at fault, if one person hears it, it is the other end at fault. ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] 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 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 issue to ground on whats the source. Its not just the phone or only the network but may be both. I am not sure how Asterisk would contribute to this. At time for a given 2 internal extension there was no echo but suddenly turned up. People dialing on my phone have echo but not on other at the same time I have few phones which I dial & no echo. So ya dont know whats wrong. Thanks all for your inputs & sharing ur experience. -- Deepak Darryl Dunkin <ddunkin@netos.net> wrote: This should only be for TDM to TDM calls, SIP to SIP calls don't use the zaptel driver. ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] 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. In your zapata.conf file try changing: echocancelwhenbridged=no to: echocancelwhenbridged=yes _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ________________________________ What kind of emailer are you? Find out today - get a free analysis of your email personality. Take the quiz at the Yahoo! Mail Championship <http://uk.rd.yahoo.com/mail/uk/taglines/default/championships/quiz/*htt p://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk/> . -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070612/b58765f1/attachment.htm