[asterisk-users] MeetMe Limits

[Sam]

I am thinking about using my asterisk server to host a conference with 
about 12 other people from around the USA.  Bandwidth issues aside, will 
this work or will all the different latencies cause issues?  Yea I know, 
I could just "try it and find out" but it is going to take alot of time 
to get everyones schedule to line up, I don't want to go through the 
trouble if I will just be disappointed.

Thanks,

Sam

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Re: [asterisk-users] Trunk/Peering provider in Canada

[Daniel Lynes]

Sascha Ferley wrote:
> Hi, 
> 
> Anyone know any decent trunk provider in Canada that can offer multiple
> channel trunks (16channels) via IAX or Sip trunking? Having some pleasant
> experience with IAXTEL out of Denver, though they don't offer services into
> Canada. 

Hello, Sascha.

You'll probably want to try MTS/Allstream or Sasktel/Navigata.  They
both offer SIP trunking.  They also call it 'IP PRI trunking', or 'IP
trunking'.

-- 
+=+
: Westwood Village Computers  :
: http://www.westwoodvillagecomputers.com/:
+=+

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Re: [asterisk-users] Asterisk 1.6 vs 1.4?

[Steve Totaro]

On Thu, Jun 5, 2008 at 4:45 PM, Johansson Olle E <[EMAIL PROTECTED]> wrote:
>
> 5 jun 2008 kl. 20.45 skrev Michael Graves:
>
>>> I wonder why more vendors haven't adopted IAX yet?
>>
>> I expect that before major players adopt this protocol it'd need to be
>> confirmed as a standard by some form of international body. That was
>> underway, but lacking anyone to push the process along.
>
> Please note that the IAX draft is just an informational RFC, not
> anything that goes any IETF standards track or is endorsed by
> the IETF.
>
> There are many vendor-related protocols documented like that.
>
> (Said from the chan_sip corner).
>
> Cheers,
> /Olle
>

I have consulted on so many systems with poor audio, the first thing I
check is IAX or SIP.  If IAX, I move over to SIP and the calls are
prefect.

I avoid IAX at all costs, use OpenVPN, open tons of ports on your
firewall, whatever you can do to use SIP.  The only time I will use
IAX is if in some remote backwards part of the world, they have
several NATs so it is impossible to control.

Even www.iax.cc recommends using SIP.  Overhead is of little concern
with MPLS and big pipes.

Thanks,
Steve Totaro

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Re: [asterisk-users] Fax on FXS

[Jason Aarons (US)]

While not on the FXS port itself other things to look for;

1) set the fax machine to disable Super G3 (33.3k), try to force it to
use G3 (14.4k).

2) set the fax machine to Disable ECM (Error Correction Mode)

3) If PSTN is T1 check span for errors, etc.

4) Some protocols have fax-passthrough and modem-relay 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Watson
Sent: Saturday, June 07, 2008 1:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Fax on FXS

On June 7, 2008 11:37:20 am bilal ghayyad wrote:
> Hi List;
>
> What configuration needed to let my FXS send and
> receive FAX?
>

Your probably going to need to give some more details about your setup
before 
anybody can help you... theres really nothing special you need to
configure 
for an FXS port to attach a fax machine to it...

keep in mind that faxing over VoIP is extremely tricky at best, but if
your 
entire call path is TDM then you shouldn;t have much of a problem.

-- 
Matt Watson
http://www.mattgwatson.ca

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Re: [asterisk-users] fxotune question

[Matthew Fredrickson]

John Morey wrote:
> I switch the wires in lines 5-8 (i.e. reversed tip and ring) and reran
> fxotune to tune the lines.  fxotune.conf ended up looking exactly the same
> as before the change.  Since I was expecting/hopping to see a change but did
> not I switched everything back to the way it was. Is there a way to test the
> lines, using a multi-meter maybe, to tell if the tip and ring are correct or
> reversed?
> 
> After putting things back I reran fxotune to get the verbose output. It,
> foxtune.out.gz, is attached.  fxotune seems to have had a better time with

It seems that one way or another the attachment didn't go through.  Can 
you email the tarball to me directly or post it to a website?

Thanks,
Matthew Fredrickson

> line 7 during this run.  fxotune.conf now contains:
> 
>5=7,255,251,251,2,255,255,1,255
>6=7,255,251,251,2,255,255,1,255
>7=4,0,0,0,0,0,0,0,0
>8=7,255,251,251,2,255,255,1,255
>9=4,0,0,0,0,0,0,0,0
>10=5,0,0,0,0,0,0,0,0
>11=0,0,0,0,0,0,0,0,0
>12=0,0,0,0,0,0,0,0,0
> 
> I tried calling directly into the lines above and it seems lines 5,6,8 have
> much more echo than lines 7,9,10. So just for fun I edited fxotune.conf to
> the following and reloaded (fxotune -s) it:
> 
>5=5,0,0,0,0,0,0,0,0
>6=5,0,0,0,0,0,0,0,0
>7=4,0,0,0,0,0,0,0,0
>8=5,0,0,0,0,0,0,0,0
>9=4,0,0,0,0,0,0,0,0
>10=5,0,0,0,0,0,0,0,0
>11=0,0,0,0,0,0,0,0,0
>12=0,0,0,0,0,0,0,0,0
> 
> Unless I am just spacing out the echo on 5,6,8 seems less now.  I really
> have no idea what is going on.
> 
> John
> 
> 
> On Fri, Jun 6, 2008 at 1:31 PM, Matthew Fredrickson <[EMAIL PROTECTED]>
> wrote:
> 
>> John Morey wrote:
>>> Tilghman,
>>>
>>> Thanks for the pointer.  I'll check this tomorrow and let you know.
>> Also, I would like to see the output without the "-d" flag and with the
>> "-v" flag.  This will output a lot of data (the echo ratio for every
>> possible coefficient setting it has tried per port).
>>
>> Matthew Fredrickson
>>
>>> John
>>>
>>> On Wed, Jun 4, 2008 at 11:18 PM, Tilghman Lesher <
>>> [EMAIL PROTECTED]> wrote:
>>>
 On Wednesday 04 June 2008 22:02:19 John Morey wrote:
> Hello,
>
> I've run fxotune at different times but continue to get what seem to be
> strange numbers in /etc/fxotune.conf.  It ends up with:
>
> 5=7,255,251,251,2,255,255,1,255
> 6=7,255,251,251,2,255,255,1,255
> 7=7,255,251,251,2,255,255,1,255
> 8=9,2,250,253,4,252,0,255,255
> 9=4,0,0,0,0,0,0,0,0
> 10=5,0,0,0,0,0,0,0,0
> 11=0,0,0,0,0,0,0,0,0
> 12=0,0,0,0,0,0,0,0,0
> ports 5-10 have lines hooked up to them.  The first four lines seem
 strange
> when compaired to what others have posted and what ports 9 and 10 have.
>
> Also if I'm reading things right my echo ratios seem to be very
> high.  Running "fxotune -d -b 5 -w 1004" gives the following:
> Dumping module /dev/zap/5
> echo ratio = 0.1759 (1960.0 / 11145.0)
> Which I read to be over 17%.  This seems crazy.  Am I reading this
>> right?
> Where should I start to look for problems?
 You might check to see if the tip and ring are reversed in your wiring.
  That
 can frequently cause weird echo problems.

 --
 Tilghman

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>>>
>>> 
>>>
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>>> asterisk-users mailing list
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>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> --
>> Matthew Fredrickson
>> Software/Firmware Engineer
>> Digium, Inc.
>>
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
> 
> 
> 
> 
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Software/Firmware Engineer
Digium, Inc.

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Re: [asterisk-users] PoE budget

[Jerry Jones]

On Jun 7, 2008, at 9:51 AM, Rob Hillis wrote:

>> On the Linksys side, we have a load of SRW-224P switches out in  
>> the wild powering 24 Snom 370s (around 7W each) off each switch.
>>
>>
>>
>
> Likewise, we sell these things by the bucket load and have no problems
> powering phones from all 24 ports.


Just curious - have these ever gotten quieter? We installed one when  
they first came out and it was WAY to loud for an office environment,  
data center would be OK.

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Re: [asterisk-users] Question on DeadAGI

[Nhadie Ramos]

Thanks to all the help. I think i have it now. I reset the CDR on the hangup 
channel.

[macro-dialout-trunk]

exten => s,1,Wait(1)

exten => s,n,Dial(SIP/[EMAIL PROTECTED],30,t)

exten => s,n,Hangup()

exten => h,1,ResetCDR(w)

exten => h,n,NoCDR()

exten => h,n,DEADAGI(get-total.php)



AGI Rx << EXEC Noop ROWCOUNT=1
    -- AGI Script Executing Application: (Noop) Options: 
(ROWCOUNT=1)
AGI Tx >> 200 result=0
AGI Rx << EXEC Noop BILLSEC=21
    -- AGI Script Executing Application: (Noop) Options: 
(BILLSEC=21)

now i can see my billsec. thanks again for all the help.

regards,
nhadie
--- On Sat, 6/7/08, Nhadie Ramos <[EMAIL PROTECTED]> wrote:
From: Nhadie Ramos <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] Question on DeadAGI
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Date: Saturday, June 7, 2008, 10:39 PM

Hi Sir,

I tried it this way, and now i can see my DEADGI being called next prob is 
on  that script i query the cdr table with the uniqueid. tried counting 
the row result first , and result was 0.

how can i make sure that it was already at the CDR table before i call my agi? 
i tried to use ResetCDR() and also without ResetCDR() but still 0 result on the 
row.

but when i query manully on the mysql console, i can see the cll was logged.

Thank You
[macro-dialout-trunk]
exten => s,1,Wait(1)
exten => s,n,Dial(SIP/[EMAIL PROTECTED],30,t)
exten => s,n.ResetCDR()
exten => s,n,Hangup
exten => h,1,DEADAGI(get-total.php)


--- On Sat, 6/7/08, Lenz
 <[EMAIL PROTECTED]> wrote:
From: Lenz <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] Question on DeadAGI
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Date: Saturday, June 7, 2008, 12:50 PM

You should use it on the hang-up extension and only after the channel is  
technically dead.
It works fine for that.
l.




On Sat, 07 Jun 2008 01:25:37 +0200, Nhadie Ramos <[EMAIL PROTECTED]>
 
wrote:

> Hi,
>
> How can i get the deadAGI to work at this scenario
>
> Basically when someonc calls international,  i will get the  
> remaining balance using AGI get-available.php.
>
> but after the call i would like to get the usage by calling  
> get-usage.php so i
 can update users balance, but looking at the debug it  
> seems the AGI was not called. is there som
>
> exten => _00.,1,AGI(get-available.php)
> exten => _00.,n,GotoIf($["${CALLSTATUS}" =
"1"]?70)
> exten => _00.,n,GotoIf($["${CALLSTATUS}" =
"2"]?80)
> exten => _00.,70,Dial(SIP/[EMAIL PROTECTED])
> exten => _00.,n,Hangup
> exten => _00.,n,DEADAGI(get-usage.php)
> exten => _00.,80,Busy
> exten => _00.,n,Hangup
>
>
> Regards,
> Nhadie
>
>
>



-- 
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Re: [asterisk-users] Manager Originate CDR problem

[Jan Eirik Sandnes]

Actually, i do get CDR on the originate, but NOT on the Dial() in the
context provided in the originate.

It looks like this:

[callgw]
exten => _X.,1,Set(CALLERID(num)=1123)
exten => _X.,n,Set(CALLERID(name)="John Travolta")
exten => _X.,n,Set(CDR(userfield)=${SUBSCR})
;Tried with answer to see if that fixed the problem
;exten => _X.,n,Answer
exten => _X.,n,Dial(SIP/195.219.218.153/${EXTEN})


On Sun, Jun 8, 2008 at 12:37 AM, Jan Eirik Sandnes <[EMAIL PROTECTED]>
wrote:

> Hello everyone!
>
> I need some help here, we have a callback service, which we use the
> Originate function to create the two calls between "A" and "B",
>
> at first we call the a leg, with originate, and then, we have a context
> which trig another dial to the B leg,
> the cdr i get is the B leg cdr, not the A leg CDR, but i really need them
> both.
>
> Is there by any chanse a workaround for this? I have read about asterisk
> and originate, and have found out that asterisk do not
> create cdrs for originate.
>
> We are running the latest asterisk(1.4.20), i need to have this fixed by
> monday :-P
>
> --
> Jan Eirik Sandnes




-- 
Jan Eirik Sandnes
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Re: [asterisk-users] Question on DeadAGI

[Nhadie Ramos]

Hi Sir,

I tried it this way, and now i can see my DEADGI being called next prob is 
on  that script i query the cdr table with the uniqueid. tried counting 
the row result first , and result was 0.

how can i make sure that it was already at the CDR table before i call my agi? 
i tried to use ResetCDR() and also without ResetCDR() but still 0 result on the 
row.

but when i query manully on the mysql console, i can see the cll was logged.

Thank You
[macro-dialout-trunk]
exten => s,1,Wait(1)
exten => s,n,Dial(SIP/[EMAIL PROTECTED],30,t)
exten => s,n.ResetCDR()
exten => s,n,Hangup
exten => h,1,DEADAGI(get-total.php)


--- On Sat, 6/7/08, Lenz <[EMAIL PROTECTED]> wrote:
From: Lenz <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] Question on DeadAGI
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Date: Saturday, June 7, 2008, 12:50 PM

You should use it on the hang-up extension and only after the channel is  
technically dead.
It works fine for that.
l.




On Sat, 07 Jun 2008 01:25:37 +0200, Nhadie Ramos <[EMAIL PROTECTED]>
 
wrote:

> Hi,
>
> How can i get the deadAGI to work at this scenario
>
> Basically when someonc calls international,  i will get the  
> remaining balance using AGI get-available.php.
>
> but after the call i would like to get the usage by calling  
> get-usage.php so i can update users balance, but looking at the debug it  
> seems the AGI was not called. is there som
>
> exten => _00.,1,AGI(get-available.php)
> exten => _00.,n,GotoIf($["${CALLSTATUS}" =
"1"]?70)
> exten => _00.,n,GotoIf($["${CALLSTATUS}" =
"2"]?80)
> exten => _00.,70,Dial(SIP/[EMAIL PROTECTED])
> exten => _00.,n,Hangup
> exten => _00.,n,DEADAGI(get-usage.php)
> exten => _00.,80,Busy
> exten => _00.,n,Hangup
>
>
> Regards,
> Nhadie
>
>
>



-- 
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http://queuemetrics.com

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[asterisk-users] Manager Originate CDR problem

[Jan Eirik Sandnes]

Hello everyone!

I need some help here, we have a callback service, which we use the
Originate function to create the two calls between "A" and "B",

at first we call the a leg, with originate, and then, we have a context
which trig another dial to the B leg,
the cdr i get is the B leg cdr, not the A leg CDR, but i really need them
both.

Is there by any chanse a workaround for this? I have read about asterisk and
originate, and have found out that asterisk do not
create cdrs for originate.

We are running the latest asterisk(1.4.20), i need to have this fixed by
monday :-P

-- 
Jan Eirik Sandnes
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Re: [asterisk-users] Bad ringback tone on zap channel

[James Lamanna]

Hmm ok.
This was a call from a SIP phone registered with Asterisk outbound on
a Zap trunk.
So would Asterisk or the phone be generating the ringback tone in that case?

It also happens very intermittently (maybe 1 in 10 calls at most...)

-- James

Rob Hillis wrote:
> In my experience, the ringback you get over a zap channel (be it
> analogue or digital) is generated by the remote end, /not/ Zaptel.
>
> The ringback you get over a SIP or IAX2 channel is often generated by
> either Asterisk or the SIP/IAX2 device you're calling from.
>
>
> James Lamanna wrote:
>> Hi,
>> I've noticed that sometimes instead of getting a regular ring tone
>> when calling out on a Zap channel, I get this obnoxious loud noise
>> which forces me to hang up.
>> Is this a problem in the Zaptel driver? I seem to recall that ringback
>> tones are generated by zaptel when dialing out from a SIP phone over a
>> Zap trunk.

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Re: [asterisk-users] Question on DeadAGI

[Nhadie Ramos]

Hi sir,

i'm sorry but how can i use it on a hangup channel?

>> exten => _00.,1,AGI(get-available.php)
>> exten => _00.,n,GotoIf($["${CALLSTATUS}" = "1"]?70)
>> exten => _00.,n,GotoIf($["${CALLSTATUS}" = "2"]?80)
>> exten => _00.,70,Dial(SIP/[EMAIL PROTECTED])
>> exten => _00.,n,Hangup
>> exten => h,n,DEADAGI(get-usage.php)   
<---should i do it like this?
>> exten => _00.,80,Busy
>> exten => _00.,n,Hangup

thank you


--- On Sat, 6/7/08, Lenz <[EMAIL PROTECTED]> wrote:
From: Lenz <[EMAIL PROTECTED]>
Subject: Re: [asterisk-users] Question on DeadAGI
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Date: Saturday, June 7, 2008, 12:50 PM

You should use it on the hang-up extension and only after the channel is  
technically dead.
It works fine for that.
l.




On Sat, 07 Jun 2008 01:25:37 +0200, Nhadie Ramos <[EMAIL PROTECTED]>
 
wrote:

> Hi,
>
> How can i get the deadAGI to work at this scenario
>
> Basically when someonc calls international,  i will get the  
> remaining balance using AGI get-available.php.
>
> but after the call i would like to get the usage by calling  
> get-usage.php so i can update users balance, but looking at the debug it  
> seems the AGI was not called. is there som
>
> exten => _00.,1,AGI(get-available.php)
> exten => _00.,n,GotoIf($["${CALLSTATUS}" =
"1"]?70)
> exten => _00.,n,GotoIf($["${CALLSTATUS}" =
"2"]?80)
> exten => _00.,70,Dial(SIP/[EMAIL PROTECTED])
> exten => _00.,n,Hangup
> exten => _00.,n,DEADAGI(get-usage.php)
> exten => _00.,80,Busy
> exten => _00.,n,Hangup
>
>
> Regards,
> Nhadie
>
>
>



-- 
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http://queuemetrics.com

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Re: [asterisk-users] Asterisk 1.6 vs 1.4?

[Gavin Henry]

2008/6/7 Gavin Henry <[EMAIL PROTECTED]>:
> What model in the Polycom or Aastra range is the 360 level with?

Probably the IP601:

http://www.voipon.co.uk/polycom-soundpoint-ip601-p-121.html

and 57i:

http://www.voipon.co.uk/aastra-57i-ip-phone-p-420.html

Snom 360:

http://www.voipon.co.uk/snom-360-ip-telephone-p-31.html

They all have the 12 keys.

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Re: [asterisk-users] Asterisk 1.6 vs 1.4?

[Gavin Henry]

What model in the Polycom or Aastra range is the 360 level with?

2008/6/6 Chris Bagnall <[EMAIL PROTECTED]>:
>> When I pushed some vendors for prices there was only a tiny gap between
>> the 300 and 360.  Would suggest looking hard at the 360 always...
>
> Interesting... here in the UK the price difference between the 300 and 360 is 
> pretty huge. Either you're getting some stunningly good pricing on 360s or 
> some abysmal pricing on the 300s :-)
>
> Regards,
>
> Chris
>
>
>
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Re: [asterisk-users] MiixMonitor filename for queue calls.

[Kevin Smith]

Hi Ed,

Glad to see you figured out your problem. I'm not sure what the 
differences are between your config and mine, but maybe this will help 
others too.

I add and remove my agents from the queue. So my agents.conf file is 
just the presistentagens=yes. Also I just run the command in the dial 
plan like below which saved mine items just fine. No configurations in 
the queue.conf file for the monitor type.

exten => 852,n,MixMonitor(/mercury/recordings/holding/${UNIQUEID}.gsm|b|)

 From there, in the hangup extension, I run a php script to take the CDR 
record and the file (rename it of course to 
queue-extension-callerid-callid-timestamp.gsm), and place it into the 
agents folder and the database for our agents/supervisors to review or 
download them.

Kevin


Ed Nunez wrote:
>
> Can anyone give me input on the following issue?
>
>  
>
> I have a queue with MixMonitor enabled. 
>
> This is also enabled in agents.conf.  
>
> On my extensions.conf, I am setting the monitor filename as fillows, 
> although I see the filename as desired in the console as I make my 
> test call, the system is only using the default file name to save the 
> mixmonitor file   (agented + uniqueID)
>
>  
>
> Agents.conf
>
>  
>
> [general]
>
> persistentagents=yes
>
>  
>
> [agents]
>
> maxlogintries=3
>
> musiconhold => default
>
> updatecdr=yes
>
> recordagentcalls=yes
>
> recordformat=wav49
>
> urlprefix=http://pbx.netoneint.com/calls/
>
> savecallsin=/var/calls
>
>  
>
> agent => 1000,1000,Ed Test1
>
> agent => 1001,1001,Ed Test2
>
>  
>
>  
>
> queues.conf
>
>  
>
> [noi-noc]  
>
> monitor-format = wav49  
>
> monitor-type = MixMonitor  
>
>  
>
> member => Agent/1001
>
> member => Agent/1000
>
>  
>
>  
>
> extensions.conf
>
>  
>
> exten => 
> 8484,1,Set(MONITOR_FILENAME=QUEUE-NOC-${CALLERID(NUM)}-${STRFTIME(${EPOCH)
>
> exten => 8484,1,answer
>
> exten => 8484,2,Queue(noi-noc)
>
>  
>
>  
>
> Console output
>
>  
>
> -- Executing [EMAIL PROTECTED]:1] Set("Zap/1-1", 
> "MONITOR_FILENAME=QUEUE-NOC-4073844200-Fri Jun  6 15:06:38 2008") in 
> new stack
>
> -- Executing [EMAIL PROTECTED]:2] Queue("Zap/1-1", "noi-noc") in 
> new stack
>
> -- Started music on hold, class 'default', on Zap/1-1
>
> -- outgoing agentcall, to agent '1001', on 
> 'Local/[EMAIL PROTECTED],1'
>
> -- Called Agent/1001
>
> -- Executing [EMAIL PROTECTED]:1] 
> Dial("Local/[EMAIL PROTECTED],2", "SIP/1658") in new stack
>
> -- Called 1658
>
> -- SIP/1658-087e7610 is ringing
>
> -- Agent/1001 is ringing
>
> -- SIP/1658-087e7610 answered Local/[EMAIL PROTECTED],2
>
> -- Agent/1001 answered Zap/1-1
>
> -- Stopped music on hold on Zap/1-1
>
> [Jun  6 15:06:40] WARNING[3976]: app_queue.c:3014 try_calling: The 
> device state of this queue member, Agent/1001, is still 'Not in Use' 
> when it probably should not be! Please check UPGRADE.txt for correct 
> configuration settings.
>
>   == Begin MixMonitor Recording Zap/1-1
>
>   == Spawn extension (numberplan-custom-3, 1658, 1) exited non-zero on 
> 'Local/[EMAIL PROTECTED],2'
>
>   == Spawn extension (incoming-att, 8484, 2) exited non-zero on 'Zap/1-1'
>
> -- Hungup 'Zap/1-1'
>
>   == End MixMonitor Recording Zap/1-1
>
>  
>
>  
>
>  
>
>  
>
>  
>
> 
>
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-- 
Kevin Smith

--- 
Mercury Network
Technical Support
Phone: 989.837.3790
Toll Free: 888.866.4638
www.mercury.net


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Re: [asterisk-users] fxotune question

[John Morey]

I switch the wires in lines 5-8 (i.e. reversed tip and ring) and reran
fxotune to tune the lines.  fxotune.conf ended up looking exactly the same
as before the change.  Since I was expecting/hopping to see a change but did
not I switched everything back to the way it was. Is there a way to test the
lines, using a multi-meter maybe, to tell if the tip and ring are correct or
reversed?

After putting things back I reran fxotune to get the verbose output. It,
foxtune.out.gz, is attached.  fxotune seems to have had a better time with
line 7 during this run.  fxotune.conf now contains:

   5=7,255,251,251,2,255,255,1,255
   6=7,255,251,251,2,255,255,1,255
   7=4,0,0,0,0,0,0,0,0
   8=7,255,251,251,2,255,255,1,255
   9=4,0,0,0,0,0,0,0,0
   10=5,0,0,0,0,0,0,0,0
   11=0,0,0,0,0,0,0,0,0
   12=0,0,0,0,0,0,0,0,0

I tried calling directly into the lines above and it seems lines 5,6,8 have
much more echo than lines 7,9,10. So just for fun I edited fxotune.conf to
the following and reloaded (fxotune -s) it:

   5=5,0,0,0,0,0,0,0,0
   6=5,0,0,0,0,0,0,0,0
   7=4,0,0,0,0,0,0,0,0
   8=5,0,0,0,0,0,0,0,0
   9=4,0,0,0,0,0,0,0,0
   10=5,0,0,0,0,0,0,0,0
   11=0,0,0,0,0,0,0,0,0
   12=0,0,0,0,0,0,0,0,0

Unless I am just spacing out the echo on 5,6,8 seems less now.  I really
have no idea what is going on.

John


On Fri, Jun 6, 2008 at 1:31 PM, Matthew Fredrickson <[EMAIL PROTECTED]>
wrote:

> John Morey wrote:
> > Tilghman,
> >
> > Thanks for the pointer.  I'll check this tomorrow and let you know.
>
> Also, I would like to see the output without the "-d" flag and with the
> "-v" flag.  This will output a lot of data (the echo ratio for every
> possible coefficient setting it has tried per port).
>
> Matthew Fredrickson
>
> > John
> >
> > On Wed, Jun 4, 2008 at 11:18 PM, Tilghman Lesher <
> > [EMAIL PROTECTED]> wrote:
> >
> >> On Wednesday 04 June 2008 22:02:19 John Morey wrote:
> >>> Hello,
> >>>
> >>> I've run fxotune at different times but continue to get what seem to be
> >>> strange numbers in /etc/fxotune.conf.  It ends up with:
> >>>
> >>> 5=7,255,251,251,2,255,255,1,255
> >>> 6=7,255,251,251,2,255,255,1,255
> >>> 7=7,255,251,251,2,255,255,1,255
> >>> 8=9,2,250,253,4,252,0,255,255
> >>> 9=4,0,0,0,0,0,0,0,0
> >>> 10=5,0,0,0,0,0,0,0,0
> >>> 11=0,0,0,0,0,0,0,0,0
> >>> 12=0,0,0,0,0,0,0,0,0
> >>> ports 5-10 have lines hooked up to them.  The first four lines seem
> >> strange
> >>> when compaired to what others have posted and what ports 9 and 10 have.
> >>>
> >>> Also if I'm reading things right my echo ratios seem to be very
> >>> high.  Running "fxotune -d -b 5 -w 1004" gives the following:
> >>> Dumping module /dev/zap/5
> >>> echo ratio = 0.1759 (1960.0 / 11145.0)
> >>> Which I read to be over 17%.  This seems crazy.  Am I reading this
> right?
> >>> Where should I start to look for problems?
> >> You might check to see if the tip and ring are reversed in your wiring.
> >>  That
> >> can frequently cause weird echo problems.
> >>
> >> --
> >> Tilghman
> >>
> >> ___
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> >>
> >
> >
> > 
> >
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>
>
> --
> Matthew Fredrickson
> Software/Firmware Engineer
> Digium, Inc.
>
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Re: [asterisk-users] Fax on FXS

[Tzafrir Cohen]

On Sat, Jun 07, 2008 at 08:37:20AM -0700, bilal ghayyad wrote:
> Hi List;
> 
> What configuration needed to let my FXS send and
> receive FAX?

Technically nothing. A fax behaves just like a normal phone device.

What you should be careful about is the path from the PSTN to that FXS
port.

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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Re: [asterisk-users] Fax on FXS

[Matt Watson]

On June 7, 2008 11:37:20 am bilal ghayyad wrote:
> Hi List;
>
> What configuration needed to let my FXS send and
> receive FAX?
>

Your probably going to need to give some more details about your setup before 
anybody can help you... theres really nothing special you need to configure 
for an FXS port to attach a fax machine to it...

keep in mind that faxing over VoIP is extremely tricky at best, but if your 
entire call path is TDM then you shouldn;t have much of a problem.

-- 
Matt Watson
http://www.mattgwatson.ca

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Re: [asterisk-users] Help-ASTERISK-MFCR2

[Moises Silva]

You need to enable loglevel=255 in unicall.conf and enable all the
levels of logging in logger.conf, otherwise the logs you post don't
say much.

Moisés Silva

On Fri, Jun 6, 2008 at 2:58 PM, Mariano Borgognone
<[EMAIL PROTECTED]> wrote:
>
> Dears,
> I have problem ASTERISK with PSTN SIEMENS EWSD (MFC R2), I don´t receive
> call for PSTN, i don´t understand why. please i need your help 
>
> # MFC/R2 normalmente no usa CRC4
> span=1,1,0,cas,hdb3
> cas=1-15:1101
> dchan=16
> cas=17-31:1101
> loadzone=us
> defaultzone=us
>
>
>  [channels]
> usecallerid=yes
> hidecallerid=no
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> echotraining=yes
> rxgain=0.0
> txgain=0.0
> group=1
> callgroup=1
> pickupgroup=1
> immediate=no
> musiconhold=default
> protocolclass=mfcr2
> protocolvariant=ar,10,10
> protocolend=cpe
> group = 1
> context= e1-incoming
> channel => 1-15
> channel => 17-31
> ;skip time slot 16
>
>
>
> Here is the LOGS when I try do make calls
>
> Jun  6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1  <- 0001  [1/   1/Idle  /Idle ]
> Jun  6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1 Detected
> Jun  6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1 Making a new call with CRN 32769
> Jun  6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1 1101  ->  [2/   2/Idle  /Idle ]
> Jun  6 16:02:18 WARNING[5060]: chan_unicall.c:2644 handle_uc_event:
> Unicall/1 event Detected
> Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1  <- 1001  [2/   2/Seize ack /Seize ack]
> Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1 Far end disconnected(cause=Normal, unspecified cause [31]) - state
> 0x2
> Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:2644 handle_uc_event:
> Unicall/1 event Far end disconnected
> Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:2930 handle_uc_event: CRN
> 32769 - far disconnected cause=Normal, unspecified cause [31]
> Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1 Call control(6)
> Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1 Drop call(cause=Normal Clearing [16])
> Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1 Call disconnected(cause=Normal, unspecified cause [31]) - state
> 0x800
> Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:2644 handle_uc_event:
> Unicall/1 event Drop call
> Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1 Call control(7)
> Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1 Release call
> Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1 1001  ->  [1/1000/Clear fwd /Seize ack]
> Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1 Release guard expired
> Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1 Destroying call with CRN 32769
> Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:2644 handle_uc_event:
> Unicall/1 event Release call
> -- Unicall/1 released
> Jun  6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2
> UniCall/1 Channel echo cancel
>
> Best Regards,
> Mariano Borgognone
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-- 
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[asterisk-users] Fax on FXS

[bilal ghayyad]

Hi List;

What configuration needed to let my FXS send and
receive FAX?

Regards
Bilal


  

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Re: [asterisk-users] fxotune vs rxgain/txgain

[Matthew Fredrickson]

Noah Miller wrote:
> Hi Matthew -
> 
>> These techniques are not mutually exclusive, I usually want people to
>> use gain modification as the last step in trying to eliminate echo
>> (after balancing the hybrid and making sure you are using a good echo
>> canceller).
>>
>> In the case of running fxotune, your zapata.conf software gain levels
>> should not affect its operation.  If you are using any of the hardware
>> gain settings (wctdm24xxp module parameters) you should normalize those
>> to 0 beforehand so that they do not interfere with the calibration process.
> 
> Thanks for your responses!
> 
> I actually didn't realize there are hardware gain settings available
> for wctdm24xxp (is there any documentation on this?  I can't seem to
> find any).  I assume the hardware gains default to 0 if left unset?

Correct.  They are set as module parameters, and actually only apply to 
fxo modules.

> Just two more questions:
> 1) I think we were experiencing ECFO with an rxgain setting of +10db
> (after having balanced the hybrid using fxotune).  I'm guessing this
> is because that rxgain value amplifies the echo a bit too much.  I
> know this is a bit of a loaded question, but is there a certain range
> of values for rxgain/txgain that we should stay within in order to
> avoid exacerbating any echo issues?

I couldn't give you exact numbers off the top of my head.  It's not hard 
to notice though if it's happening :-)

> 2) Are rxgain/txgain values applied before or after hardware echo 
> cancellation?

rxgain is pre-hardware echo canceller and txgain is post hardware echo 
canceller. (zapata.conf rxgain and txgain).

-- 
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.

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Re: [asterisk-users] PoE budget

[Rob Hillis]

Chris Bagnall wrote:
>> have used many fsm7326p to power 24 phones or 726tp to power 12
>> phones and they work great
>> 
>
> On the Linksys side, we have a load of SRW-224P switches out in the wild 
> powering 24 Snom 370s (around 7W each) off each switch.
>
>
>   

Likewise, we sell these things by the bucket load and have no problems 
powering phones from all 24 ports.

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Re: [asterisk-users] Bad ringback tone on zap channel

[Rob Hillis]

In my experience, the ringback you get over a zap channel (be it 
analogue or digital) is generated by the remote end, /not/ Zaptel.

The ringback you get over a SIP or IAX2 channel is often generated by 
either Asterisk or the SIP/IAX2 device you're calling from.


James Lamanna wrote:
> Hi,
> I've noticed that sometimes instead of getting a regular ring tone
> when calling out on a Zap channel, I get this obnoxious loud noise
> which forces me to hang up.
> Is this a problem in the Zaptel driver? I seem to recall that ringback
> tones are generated by zaptel when dialing out from a SIP phone over a
> Zap trunk.
>
> Thanks.
>
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>   

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Re: [asterisk-users] features.conf not working

[Michiel van Baak]

On 08:36, Sat 07 Jun 08, Russell Bryant wrote:
> 
> On Jun 6, 2008, at 4:33 PM, Manolet Gmail wrote:
> > i have this on my features.conf:
> >
> > [applicationmap]
> > testfeature => *9,callee,Playback,tt-monkeys
> >
> > extensions.conf:
> >
> > [globals]
> > DYNAMIC_FEATURES=testfeature
> > trunk_1 = Zap/g1
> > trunk_2 = Zap/g2
> >
> >
> > what else i have to add in order to make this works? im using 2 xlite,
> 
> Just a hunch ... if you're using xltite, it's likely that you're not  
> pressing the digits fast enough to satisfy the default timeout.  The  
> default "featuredigittimeout" is 500 ms.  Change this option in  
> features.conf and increase it to 2000 ms and try again.

another tip:
Make sure you have the dtmfmode for the xlite sip stanza set correctly.

-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer aficionados are both called users?"


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Re: [asterisk-users] features.conf not working

[Russell Bryant]

On Jun 6, 2008, at 4:33 PM, Manolet Gmail wrote:
> i have this on my features.conf:
>
> [applicationmap]
> testfeature => *9,callee,Playback,tt-monkeys
>
> extensions.conf:
>
> [globals]
> DYNAMIC_FEATURES=testfeature
> trunk_1 = Zap/g1
> trunk_2 = Zap/g2
>
>
> what else i have to add in order to make this works? im using 2 xlite,

Just a hunch ... if you're using xltite, it's likely that you're not  
pressing the digits fast enough to satisfy the default timeout.  The  
default "featuredigittimeout" is 500 ms.  Change this option in  
features.conf and increase it to 2000 ms and try again.

--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.





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Re: [asterisk-users] Question on DeadAGI

[Lenz]

You should use it on the hang-up extension and only after the channel is  
technically dead.
It works fine for that.
l.




On Sat, 07 Jun 2008 01:25:37 +0200, Nhadie Ramos <[EMAIL PROTECTED]>  
wrote:

> Hi,
>
> How can i get the deadAGI to work at this scenario
>
> Basically when someonc calls international,  i will get the  
> remaining balance using AGI get-available.php.
>
> but after the call i would like to get the usage by calling  
> get-usage.php so i can update users balance, but looking at the debug it  
> seems the AGI was not called. is there som
>
> exten => _00.,1,AGI(get-available.php)
> exten => _00.,n,GotoIf($["${CALLSTATUS}" = "1"]?70)
> exten => _00.,n,GotoIf($["${CALLSTATUS}" = "2"]?80)
> exten => _00.,70,Dial(SIP/[EMAIL PROTECTED])
> exten => _00.,n,Hangup
> exten => _00.,n,DEADAGI(get-usage.php)
> exten => _00.,80,Busy
> exten => _00.,n,Hangup
>
>
> Regards,
> Nhadie
>
>
>



-- 
Loway Research - Home of QueueMetrics
http://queuemetrics.com

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Re: [asterisk-users] Disable sending CNAM over facility for 2bct

[Matt Florell]

I think this is a bit beyond the average users-list question. There
are very few people who do 2BCT and it was quite difficult to get
anyone to help last year when I was trying to get it working on NI2 in
libpri. I'm not really sure how to go about what you are asking, but I
would suggest getting on the IRC channel for Asterisk and asking
around there.

Also if you can somehow get a hold of Matt Fredrickson(who is a very
busy guy)  at Digium, he could probably figure this out in a matter of
minutes.

MATT---


On 6/6/08, Remi Quezada <[EMAIL PROTECTED]> wrote:
> Hey,
>
>  Is there a way I can disable sending cnam over the facility message when
>  I am performing a two b-channel transfer?
>
>  Thanks,
>
>  Remi
>
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Re: [asterisk-users] Call Transfer

[randulo]

On Sat, Jun 7, 2008 at 8:24 AM, Theodore Patsiouras
<[EMAIL PROTECTED]> wrote:
> If my secretary or anyone else picks up the call when the line is transferred 
> in my ext then I have the > internal caller ID. Can I have somehow the 
> External callerID?

Look at the channel variables that contain the callerid information.
You can assign the incoming callerid to the one that makes the call to
your local extension to do what you wish.

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