[Asterisk-Users] Re: Incoming echo cancel

[Jon Bebeau]

I'm having echo too - ISDN-PRI using a Sangoma card to the PSTN.  My SIP 
outgoing calls have an echo about 80% of the time, but only on a local T1. 
It only an echo to the SIP caller; the called party never hears the echo.  I 
have a second T1-PRI (port 2 of the same card) to a long distance carrier 
with no echo problem, ever.  Played around with all the echo stuff, you 
identified below.  Seems there is much ambiguity on the guidance as much of 
the echo suggestions apply only to analog lines.  It's unclear how there can 
be an echo on a T1-PRI (digital) anyway.

Jon

----- Original Message ----- 
From: "Nenad Radosavljevic" <nenadr at deltaplan.co.yu>
To: <asterisk-users at lists.digium.com>
Sent: Friday, March 11, 2005 3:53 PM
Subject: [Asterisk-Users] Re: Incoming echo cancel


> Same problem here: if call come over ISDN PRI and it is for a SIP phone 
> that equals to strong echo situation, at the SIP end. Interestingly this 
> doesn't happen on all calls but it does on 95% of them. Asterisk load at 
> that moment is insignificant - 1 to 2 calls.
>
> I have tried with all possible echo cancellers in zconfig.h, with and 
> without MMX, and with and without CFLAGS+=-march=i686 in zaptel Makefile, 
> but without any success.
>
> TE110P is on its own interrupt - no other cards on board except Intel 
> chipset PCI LAN card (on shared interrupt) and machine is Intel 685 
> chipset based, with P4 Celleron 2.6 GHz, 1Gbyte RAM, WD 10K RPM IDE HDD.
>
> I have even contacted Digium support with this issue, but except a request 
> for some additional explanations of my setup, nothing from them so far 
> (for about a week).
>
> Anyone have an idea, why this type of echo happens ? As far as I have read 
> on the lists this type of echo should not occur at all, but it simply does 
> !
>
> Regards,
>            Nenad Radosavljevic
>
>> Hi,
>>
>>
>>  I'm having the same problems in echo cancellations that are mentioning
>> in this mail of the list
>> http://lists.digium.com/pipermail/asterisk-users/2003-July/016073.html ,
>> but I haven't found some reply to this mail.
>>
>> I haven't echo problem on outcoming calls but echo cancellation is
>> disabled in zaptel channels in incoming calls. Status of zaptel channel
>> is the next:
>>
>> localhost*CLI> zap show channel 32
>> Channel: 32LI>
>> File Descriptor: 49
>> Span: 2
>> Extension: 958238500
>> Dialing: no
>> Context: incoming
>> Caller ID string: 685975350
>> Destroy: 0
>> InAlarm: 0
>> Signalling Type: PRI Signalling
>> Owner: Zap/32-1
>> Real: Zap/32-1
>> Callwait: <None>
>> Threeway: <None>
>> Confno: -1
>> Propagated Conference: -1
>> Real in conference: 0
>> DSP: yes
>> Relax DTMF: no
>> Dialing/CallwaitCAS: 0/0
>> Default law: alaw
>> Fax Handled: no
>> Pulse phone: no
>> Echo Cancellation: 256 taps, currently OFF
>> PRI Flags: Call
>> PRI Logical Span: Implicit
>> Actual Confinfo: Num/0, Mode/0x0000
>> Actual Confmute: No
>> Actual Hookstate: Onhook
>>
>>
>>
>> I don't know because Asterisk doesn't enable echo cancelation.
>>
>>
>>
>>
>> Roberto Vargas.
>>
>>
>>
>> ------------------------------
>>
>> Message: 11
>> Date: Fri, 11 Mar 2005 13:56:26 +0200
>> From: Herman Cremer <herman at etel.co.za>
>> Subject: Re: [Asterisk-Users] how do i get rid of this blasted echo
>> !!!
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> <asterisk-users at lists.digium.com>
>> Message-ID: <1110542186.2681.14.camel at localhost.localdomain>
>> Content-Type: text/plain
>>
>> Thanks Error.
>>
>> I have switched to IAX looong ago....much better !
>> Just battle when doing double NAT :)
>>
>> I dont have the phones here with me,
>> but lets say its different...is there away
>> to adjust the channel to fix the err ?
>>
>> -herman
>>
>>
>>
>> On Fri, 2005-03-11 at 13:24, support at biz4web.com wrote:
>>> Hi Herman,
>>>
>>> Look at the bottom of your phones and compare the REN values of both. Do
>>> they both value of REN 1.0?  I think the one with the problem might have
>>> an REN value other than one.  You tell me!
>>>
>>> Errol Samuels
>>> "Don't let SIP Drive you crazy, use IAX2"
>>>
>>>
>>>
>>> > On the echo...
>>> >
>>> > I have 2 extensions, with different analog phones.
>>> > The one works fine, the other echos and scratches
>>> > like mad !!
>>> >
>>> > I have switched the ports, cables etc but its ALWAYS
>>> > the same phone...
>>> >
>>> > Maybe this could be it ?
>>> >
>>> > Is it ok from a SIP phone ?
>>> >
>>> > Herman cremer
>>> >
>>> >
>>> >
>>> >
>>> >
>>> > _______________________________________________
>>> > Asterisk-Users mailing list
>>> > Asterisk-Users at lists.digium.com
>>> > http://lists.digium.com/mailman/listinfo/asterisk-users
>>> > To UNSUBSCRIBE or update options visit:
>>> >    http://lists.digium.com/mailman/listinfo/asterisk-users
>>> >
>>>
>>>
>>> _______________________________________________
>>> Asterisk-Users mailing list
>>> Asterisk-Users at lists.digium.com
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>> To UNSUBSCRIBE or update options visit:
>>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>>
>> ------------------------------
>>
>> Message: 12
>> Date: Fri, 11 Mar 2005 13:01:05 +0000
>> From: Niksa Baldun <niksa.baldun at lumiss.hr>
>> Subject: Re: [Asterisk-Users] Unable to create Zap channel when
>> dialing using a bri cellular gateway
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> <asterisk-users at lists.digium.com>
>> Message-ID: <42319691.4050904 at lumiss.hr>
>> Content-Type: text/plain; charset=ISO-8859-1
>>
>> Obviously, your ISDN gateway is misconfigured somehow. I would suggest
>> that you configure the gateway to dial some extension on your * box and
>> see if incoming calls work. If they don't, then there is a problem with
>> configuration of gateway's ISDN interface. If incoming calls work, then
>> it is possible that the gateway is rejecting outgoing calls based on
>> number called (I had that problem once), or perhaps you just forgot to
>> pay the bill to your mobile operator :)).
>>
>> Niksa
>>
>>
>> David Masure wrote:
>>
>>>
>>>
>>> Hi all,
>>>
>>>
>>> I have an asterisk box set up with a bri card (using zaphfc).  I have
>>> a bri cellular gateway connected to it beacuse I'd like to route all
>>> my cellular calls through that gateway.
>>>
>>> The probel I encounter is that when trying to dial a phone number,
>>> I've the message : unable to create a zap channel.
>>>
>>> My card is normally well configured because when connected to the NT,
>>> It works perfectly...  The gateway is configured as a NT as well so no
>>> worry...
>>>
>>> Has anyone an idea of what I should look for ?
>>>
>>> Thank you
>>>
>>> David Masure
>>>
>>>
>>>------------------------------------------------------------------------
>>>
>>>_______________________________________________
>>>Asterisk-Users mailing list
>>>Asterisk-Users at lists.digium.com
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>> ------------------------------
>>
>> Message: 13
>> Date: Fri, 11 Mar 2005 19:58:37 +0800
>> From: Ronald Wiplinger <ronald at elmit.com>
>> Subject: [Asterisk-Users] CDR database
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> <asterisk-users at lists.digium.com>
>> Message-ID: <423187ED.2020006 at elmit.com>
>> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>>
>> I am looking at AMP and read "All the graphic & reports are based over
>> the CDR database."
>> How do I get the CDRs into a database?
>>
>>
>> bye
>>
>> Ronald
>>
>>
>>
>> ------------------------------
>>
>> Message: 14
>> Date: Fri, 11 Mar 2005 13:05:42 +0100
>> From: Marc Storck <marc.storck at msnetworks.lu>
>> Subject: [Asterisk-Users] SIP signalling and RTP to different servers
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> <asterisk-users at lists.digium.com>
>> Message-ID: <42318996.3040701 at msnetworks.lu>
>> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>>
>> Hello,
>>
>> we're in process of testing an interconnection with a trans-european
>> carrier. But the carrier wants the SIP signalling to server 1 and the
>> RTP stream to server 2. How do I configure asterisk to work with that
>> type of installation. It seems they are using NexTone as SIP Signaling
>> and RTP servers. Can someone help me???
>>
>> Regards,
>>
>> Marc
>> -- 
>> CTO                            Marc Storck
>> MS Networks SA                 mstorck at msnetworks.lu
>> IT Service Provider            http://www.msnetworks.lu
>> 15, route d'Esch               Phone: +352 2727 3030
>> L-4450 Belvaux                 Fax:   +352 2727 3060
>>
>> --------------- MS Networks powered service ---------------
>> http://www.LuxAdmin.com       Hosting and housing solutions
>> -----------------------------------------------------------
>>
>>
>>
>> ------------------------------
>>
>> Message: 15
>> Date: Fri, 11 Mar 2005 12:06:18 +0000
>> From: Robbie Hughes <spam at dynsysgroup.com>
>> Subject: RE: [Asterisk-Users] TE110P experiance
>> To: asterisk-users at lists.digium.com
>> Message-ID: <423189BA.9080609 at dynsysgroup.com>
>> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>>
>>>    - the PCI ID of the card seems to change over time which means that
>>> loading the module does not always recognise the card, only way to reset
>>> this is to power cycle the machine
>>
>> I noticed this behaviour as well.
>> i thought it was my motherboard wrongly assigning irq values -
>> the symptoms i noticed were:
>>
>> irq value set by me - machine starts, module loads, all functional
>> reboot
>> irq value reset to another /shared/ irq - machine starts, module fails
>> reboot
>> irq value set by me again -  machine starts, module fails
>> check irq - it has been reset to something else
>> irq value set by me - machine starts, module loads, all functional
>>
>> unfortunately my bios doesn't allow manual assigning of irqs - i have to
>> swap them arond based on the ones it gives me...
>> i ended up disabling my usb bus as i don't need it...
>>
>> i can't find any consistency to it and am living in fear of the reboot..
>> very odd..
>>
>>
>> ------------------------------
>>
>> Message: 16
>> Date: Fri, 11 Mar 2005 09:09:00 -0300
>> From: Renato Mintz <renatomintz at gmail.com>
>> Subject: [Asterisk-Users] SIP Phone Unreachable
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> <asterisk-users at lists.digium.com>
>> Message-ID: <c20eea0805031104097fc20222 at mail.gmail.com>
>> Content-Type: text/plain; charset=US-ASCII
>>
>> Hi Folks,
>>
>> I found a strange problem trying to install a system on a customer. I
>> have the following network configuration:
>>
>> Asterisk - Router (NAT) - Internet - Router (NAT) - Grandstream Phone
>>
>> The routers are low end D-Link router + broadband access. The router
>> near asterisk has 5060 and 10000-10009 ports opened and assigned to
>> Asterisk server. The router near the phone has default configuration:
>> outgoing ok, incoming blocked.
>>
>> I have Qualify = 1000. As soon as * is restarted I get a message
>> telling the phone is unreachable. Looking at SIP debug I see *
>> transmitting OPTIONS and receiving OK but it seems that * discards the
>> OKs, because it always transmits OPTIONS 4 times (and receives 4 OKs),
>> stop a little and begin transmitting OPTIONS again.
>>
>> Looking at the SIP messages I found that the Call-ID in the OPTIONS
>> message uses the Asterisk EXTERNAL IP address but the OK coming from
>> the GS Phone has its Call-ID with the Asterisk INTERNAL IP address.
>>
>> I run ethereal near the phone and the OK it sends has Asterisk
>> EXTERNAL IP address! Somebody is translating the EXTERNAL IP into the
>> INTERNAL IP at the Call-Id header.
>>
>> I also run tcpdump at the Asterisk Server and the result is the same
>> as the sip debug.
>>
>> My simple conclusion is: the router is opening the SIP message and
>> translating the Call-Id header IP, but I don't believe in that.
>>
>> Any clue?
>>
>> Thanks?
>>
>> Renato
>>
>>
>> ------------------------------
>>
>> Message: 17
>> Date: Fri, 11 Mar 2005 14:11:12 +0200
>> From: Yair Hakak <yhakak at gmail.com>
>> Subject: Re: [Asterisk-Users] CDR database
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> <asterisk-users at lists.digium.com>
>> Message-ID: <43ff3394050311041125ebfca3 at mail.gmail.com>
>> Content-Type: text/plain; charset=US-ASCII
>>
>> http://www.voip-info.org/wiki-Asterisk+billing
>>
>>
>> On Fri, 11 Mar 2005 19:58:37 +0800, Ronald Wiplinger <ronald at elmit.com> 
>> wrote:
>>> I am looking at AMP and read "All the graphic & reports are based over
>>> the CDR database."
>>> How do I get the CDRs into a database?
>>>
>>> bye
>>>
>>> Ronald
>>>
>>> _______________________________________________
>>> Asterisk-Users mailing list
>>> Asterisk-Users at lists.digium.com
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>> ------------------------------
>>
>> Message: 18
>> Date: Fri, 11 Mar 2005 12:15:35 -0000
>> From: "JunkMail" <junkmail at segurajuda.dyndns.org>
>> Subject: Re: [Asterisk-Users] Asterisk at home silly problem, please
>> help!
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>> <asterisk-users at lists.digium.com>
>> Message-ID: <002e01c52634$07e94490$0a00a8c0 at segurajuda.local>
>> Content-Type: text/plain; charset="iso-8859-1"
>>
>> Solved!
>> The problem was that "capiinit start" can only be done by user "root" and
>> asterisk is started as user "asterisk".
>> Once I edited sudo ("visudo") and gave permission, the problem was 
>> solved.
>>
>> Regards
>>
>> M.G.
>>
>> ----- Original Message ----- 
>> From: "Junk Mail" <junkmail at segurajuda.dyndns.org>
>> To: <asterisk-users at lists.digium.com>
>> Sent: Wednesday, March 09, 2005 11:12 PM
>> Subject: [Asterisk-Users] Asterisk at home silly problem, please help!
>>
>>
>>> Hi all!
>>>
>>> After much struggling I got my *@home working fine AND making use of two
>>> AVMFritz!PCI cards. Really nice !  (kernel 2.4.2x)
>>>
>>> There's however a silly glitch that's getting on my nerves, and, kind of 
>>> a
>>> newbie that I am to linux, it should be easy to get help :
>>>
>>> -- "capiinit start" MUST BE run before Asterisk. (any other way makes *
>> not
>>> to start because chan_capi doesn't find CAPI support)
>>>
>>> You must find this an easy thing, as I did. So I entered /etc/rc.d/ and
>>> inserted "capiinit start" to start as early as possible. Also added some
>>> lines of junk text so to see them going by as the system boots...
>>>
>>> What's making me desperate is that the lines go by, capiinit is, in 
>>> fact,
>>> runned, and Asterisk still fails in the end.
>>> I login and type my very first command "asterisk -vvvc" and it then 
>>> starts
>>> with no trouble.
>>>
>>> Is this strange or what ?
>>>
>>> Thanks in advance for your help.
>>>
>>> M.G.
>>>
>>> _______________________________________________
>>> Asterisk-Users mailing list
>>> Asterisk-Users at lists.digium.com
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>> To UNSUBSCRIBE or update options visit:
>>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>>
>> ------------------------------
>>
>> Message: 19
>> Date: Fri, 11 Mar 2005 15:12:36 +0200
>> From: Herman Cremer <herman at etel.co.za>
>> Subject: [Asterisk-Users] IAX, double NAT
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> <asterisk-users at lists.digium.com>
>> Message-ID: <1110546756.2681.22.camel at localhost.localdomain>
>> Content-Type: text/plain
>>
>> has anyone managed to get IAX client (firefly 3rd party version)
>> to work,
>> where the *Server is behind single NAT,
>> with port forwarding enabled on the NAT router, and
>> the client behind double NAT ?
>>
>> clients behind single nat to * work fine.
>>
>> hermancremer
>>
>>
>>
>> ------------------------------
>>
>> Message: 20
>> Date: Fri, 11 Mar 2005 05:12:02 -0800 (PST)
>> From: <taintedham-mailinglists at yahoo.com>
>> Subject: [Asterisk-Users] One single record file for a meetme monitor?
>> To: Asterisk-Users at lists.digium.com
>> Message-ID: <20050311131202.98140.qmail at web52507.mail.yahoo.com>
>> Content-Type: text/plain; charset=us-ascii
>>
>> I'm trying to figure out the best way to record a
>> conference.
>>
>> Many people suggest something like this:
>> exten => 2060,1,Answer
>> exten => 2060,2,Wait(1)
>> exten => 2060,3,Monitor(wav,myfilename)
>> exten => 2060,4,Meetme(1,ps)
>>
>> However, this creates two files for each user that
>> connects to the meetme.  (I know they can be mux'd
>> together to make one with sox..I've done that too)
>> However, you still get 10 files if 10 users enter the
>> meetme.
>>
>> I'd really like to be able to simple record a single
>> file with all the channels mux'd together.
>>
>> Someone suggested executing a script and having the
>> monitor application join the meetme.  However, I have
>> yet to see this work correctly.... and it isn't the
>> best solution because I've got to have some logic to
>> add the local listener when the first person enters...
>> and exit when the last person exits.
>>
>> Anyways, just wanted to see if any of you have this
>> worked out already.  I really think there should be an
>> option on the meetme.
>>
>> Thanks,
>> Dave
>>
>>
>> ------------------------------
>>
>> Message: 21
>> Date: Fri, 11 Mar 2005 13:15:00 GMT
>> From: Iqbal <iqbal at gigo.co.uk>
>> Subject: Re: [Asterisk-Users] SetCallerID({$NEWCALLERID})
>> To: asterisk-users at lists.digium.com
>> Message-ID: <o80wpBK5.1110546900.6299240.iqbal at gigo.co.uk>
>> Content-Type: text/plain; charset=ISO-8859-1
>>
>>
>> that would do it, the $ is in the wrong place
>>
>> Iqbal
>>
>> On 3/11/2005, "beonice" <beonice at yahoo.com> wrote:
>>
>>>
>>>--- Steven Frazier <lists at futuresync.com> wrote:
>>>> I am trying to SetCallerID to a variable I have
>>>> defined.  This obviously is
>>>> wrong.  It actually sets the caller ID to
>>>> $NEWCALLERID.  I have search
>>>> through the examples on wiki but wasn't able to find
>>>> something similar to
>>>> see what I was doing wrong.  Could someone tell me
>>>> the correct way to
>>>> SetCallerID to a defined variable?
>>>>
>>>>  exten => 2125551212,5,SetCallerID({$NEWCALLERID})
>>>
>>>  --- snipped the rest ---
>>>
>>>Off-hand, not having actually tested this, I'd guess
>>>that you have the $ in the wrong place. Move it one
>>>character to the left.
>>>
>>>Cheers,
>>>Maya
>>>
>>>
>>>
>>>
>>>__________________________________
>>>Do you Yahoo!?
>>>Yahoo! Small Business - Try our new resources site!
>>>http://smallbusiness.yahoo.com/resources/
>>>_______________________________________________
>>>Asterisk-Users mailing list
>>>Asterisk-Users at lists.digium.com
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>
>>
>>
>> ------------------------------
>>
>> Message: 22
>> Date: Fri, 11 Mar 2005 14:16:17 +0100
>> From: Giovanni Miano <giomiano at gmail.com>
>> Subject: [Asterisk-Users] Asterisk at home 0.6 + bristuff
>> To: Asterisk-Users at lists.digium.com
>> Message-ID: <d75be1ca05031105162038a069 at mail.gmail.com>
>> Content-Type: text/plain; charset=US-ASCII
>>
>> How install bristuff in Asterisk at home ?
>>
>> i tried version 0.2.Rca to last RC7k and when try to compile zaptel
>> (after patched it) i've this error:
>>
>> make: *** [zaptel.o] Error 1
>>
>>
>> ------------------------------
>>
>> Message: 23
>> Date: Fri, 11 Mar 2005 16:23:56 +0300 (EAT)
>> From: "Julius Kidubuka" <juki at one2net.co.ug>
>> Subject: [Asterisk-Users] Voicemail - No Audio Output!
>> To: asterisk-users at lists.digium.com
>> Message-ID: <3203.81.199.88.27.1110547436.squirrel at mail.one2net.co.ug>
>> Content-Type: text/plain;charset=iso-8859-1
>>
>> Hi all,
>>
>> I am able to receive voicemail in my mail box but when I try to play the
>> audio file attachment, I hear nothing at all (yet the caller on the other
>> end does leave a voicemail message)!
>>
>> Anyone had a similar problem before? Ideas are welcome!
>>
>> Note: I am using Asterisk at Home 0.6
>>
>> Thanks in advance,
>> -- 
>> Rgds,
>> Julius Kidubuka.
>> "My advice to you is get married: if you find a good wife you'll be 
>> happy;
>> if not, you'll become a philosopher."
>>
>>
>>
>>
>>
>>
>> ------------------------------
>>
>> Message: 24
>> Date: Fri, 11 Mar 2005 14:25:45 +0100 (CET)
>> From: Peter Svensson <psvasterisk at psv.nu>
>> Subject: RE: [Asterisk-Users] Panasonic TDA200 E1 -> E100P negotiation
>> issues
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> <asterisk-users at lists.digium.com>
>> Message-ID: <Pine.LNX.4.44.0503111403320.1941-100000 at cheetah.psv.nu>
>> Content-Type: TEXT/PLAIN; charset=US-ASCII
>>
>> On Fri, 11 Mar 2005, James Bean wrote:
>>
>>> Whooppss had pri_cpe set, redid the debug as attached.
>>>
>>> They seem the same but just in case.
>>
>> Asterisk does not see anything coming in on the D channel. What does
>> zttool say about the state of the link?
>>
>> As I said before, if the card is an isdn card you need to use ccs
>> signalling. Cas signalling is unusual, but possible, over an E1. Can you
>> find out the model number of the E1 card in the Panasonic pbx?
>>
>> Peter
>>
>>
>>
>> ------------------------------
>>
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users at lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>> End of Asterisk-Users Digest, Vol 8, Issue 89
>> *********************************************
>>
>
>
>
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