RE: RE : [Asterisk-Users] Best quad-port fxo solution with EC?

[gw]

Hello There,
Yes I have a tellabs installed, in fact I may have been one of those who helped 
you out :) 

What I need though is only 4 ports, that's a bit overkill.  I also did the spa 
and tdm400 with little luck.

Greg

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Monday, February 13, 2006 9:15 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: RE : [Asterisk-Users] Best quad-port fxo solution with EC?

In my expereince a channel bank with a digium single span card and a Tellabs EC 
perfomed the best, but is too expensive (it gives you a minimum of 8 ports). 
Next to that I use a mediatrix 1204, and compared to all others I have tried 
works best. I have tried:
Sipura SPA3000
Digium TDM400 with 4 FXO mods


On 2/13/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
 Hi,

 I have good results with the new TDM2400P serie (with the hardware 
 echocan, of course).
 May be you must check one TDM2401E to see if it's ok for you...

 Good luck.

 Best Regards,
 Francois BERGERET,
 France.


 -Message d'origine-
 De : [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] De la part de 
 [EMAIL PROTECTED] Envoyé : lundi 13 février 2006 07:36 À : 
 asterisk-users@lists.digium.com Objet : [Asterisk-Users] Best 
 quad-port fxo solution with EC?


 Hello All,

 I am trying to figure out which way to go for a quad port fxo solution 
 with a good echo can on it.  My options are the sangoma remora, a 
 mediatrix fxo, or something similar.

 The issue is that I would need a good EC.  This would be on about a 
 9000 foot loop, and the lines don't function well on a spa-3000 or 
 zaptel tdm 4 port card.

 Anyone have experience that drives them in a certain direction when 
 considering a good ec on a quad port?

 I tried this also with some fxo clones, but echo killed it.

 Thanks,
 Greg
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RE: [Asterisk-Users] Best quad-port fxo solution with EC?

[gw]

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Monday, February 13, 2006 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Best quad-port fxo solution with EC?


  I am trying to figure out which way to go for a quad port fxo 
  solution with a good echo can on it.  My options are the sangoma 
  remora, a mediatrix fxo, or something similar.
  
  The issue is that I would need a good EC.  This would be on about a 
  9000 foot loop, and the lines don't function well on a spa-3000 or 
  zaptel tdm
  4 port card.
  
  Anyone have experience that drives them in a certain direction when 
  considering a good ec on a quad port?
  
  I tried this also with some fxo clones, but echo killed it.
 
 The nearest CO my POTS line goes to is 11 miles away.  My POTS line 
 works when plugged into my TDM400P FXO port.  I DID have to fiddle 
 with the gains a bit and I still have to get rid of the last of the 
 echo, but overall it seems to work well.

Are you sure the telco is not using fiber-extended line modules?
- Not certain.

Have you measured the loss from their milliwatt generator?
- Yes, results were way off, unless I did it wrong, my gains should be
about -2 on a fttp line but the measurements suggest about +7

(The numbers would be very interesting to see since a large number of
spa3k and tdm04 users at that distance have significant EC issues.)

- I'll post when I get more details...

Greg
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RE: [Asterisk-Users] Problem with Wait() and chan_capi-cm?

[gw]

Hello Florian,
I spoke to soon, thought you were referencing something else... I have
been having a problem post 8015 build of asterisk that has been
preventing me from going up any higher...

It's an odd one too, and I narrowed it down, tested like crazy, etc...

You could see my previous post about it, :) I'll get it eventually...

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Florian
Heer
Sent: Saturday, February 11, 2006 10:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Problem with Wait() and chan_capi-cm?

[EMAIL PROTECTED] wrote:

Try build 8015.  I know its odd, but this is just like the problem I am

having...
  

Uhm... sorry if I seem a bit uninformed, but how do I get that version?

Regards, Florian.
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[Asterisk-Users] Best quad-port fxo solution with EC?

[gw]

Hello All,

I am trying to figure out which way to go for a quad port fxo solution
with a good echo can on it.  My options are the sangoma remora, a
mediatrix fxo, or something similar.

The issue is that I would need a good EC.  This would be on about a 9000
foot loop, and the lines don't function well on a spa-3000 or zaptel tdm
4 port card.

Anyone have experience that drives them in a certain direction when
considering a good ec on a quad port?

I tried this also with some fxo clones, but echo killed it.

Thanks,
Greg
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RE: [Asterisk-Users] asterisk + door opener

[gw]

Maybe do a transfer to a dedicated extension, which calls the script
with the system() command to open the door?  Or use the feature keys for
a blind transfer.  Seems like it could work.

Btw, what kind of door phone opener do you have?  I've been looking for
something similar...

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thomas
Artner
Sent: Sunday, February 12, 2006 9:39 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] asterisk + door opener

Hi!

I am new to asterisk and I'd like to know wheter the following scenario
is possible:

Someone press the Button on the door station.
The door station dials lets say the extension 333.
I  take the call on 333 and talk with the person on the door.

Now I'd like to activate the door opener by pressing some numbers on the
analog telefon.
Asterisk should now recognize that I pressed something to open the door
and should execute a script which opens the door.

My question is, is it possible to execute a script while i am talking
with the person on the door, without hanging up before?

Can anyone give me some hints where to start looking in the docu?!
I only need to know how to execute a script when I press - lets say the
* Button while i am talking.

Opening the door with a bash script is already working.

Thx very much in advance!

Tom
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[Asterisk-Users] Chan capi failing post build 8015, possible causes?

[gw]

Hello List and Armin,

I have been trying to narrow down my problems with getting chan_capi to
function properly.  It seems that anything above build 8015 causes a
segfault on dial or receive.

The problem almost seems sporadic, and is certainly related to sip or
iax channels.

As soon as I update to build 8016, the problem starts. 8015 is fine.

For example, if I direct the number right to a menu, gsm plays fine.  As
soon as I dial a digit, when it tries to connect to a sip channel the
call drops.

On more frequent occasions, asterisk will segfault.

It happens all in the first calls.

What I tried doing was a clean asterisk install, with only demos, then
installing chan-capi 0.6.4, and directing the number to the demo menu.
Call still drops...

This also happens exactly the same on two different servers, both with
eicon diva server bri cards.

Build 8016 seems to address times and dates, and I did notice that the
system will die on a gotoiftime statement, but even if I take it out
there are still problems.

At first I thought it could have to do with the monitor command, but
that was not it.  Then I noticed if I was dialing with a /B, there could
be issues too...

Any ideas?  This is quite odd, and I'd like to be able to take advantage
of the newer builds...

Also, I do not have enough experience to reverse the effects of build
8016 only, and jump to a higher build without the diffs.

This is on a debian test system, with gcc 3.3.5.

I am willing to try this on another distro, but would need advice on
which direction to go.

I finally patched 8015 for the timebomb fix, so now I can have proper
dates.

Regards,
Greg
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RE: [Asterisk-Users] Problem with Wait() and chan_capi-cm?

[gw]

Try build 8015.  I know its odd, but this is just like the problem I am
having... 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Florian
Heer
Sent: Saturday, February 11, 2006 9:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Problem with Wait() and chan_capi-cm?

Hi!

I am playing around with Asterisk and have a problem :-)
(Asterisk-version: 1.2.4, chan_capi-cm-version: 0.6.4) I have a
sip-phone at my desk and an ISDN-phone (independent of the
Asterisk-server) in my living room, when I'm not at my desk, the
sip-phone is switched off. I would like to be able to accept calls at
both phones (when available) and have Voicemail kick in if I don't
answer. The 'normal' extension would be something like this:

exten = 12345,1,Dial(SIP/me,30)
exten = 12345,2,VoiceMail(su12345)

Works fine as long as the sip-phone is available, if it is not, it is
flagged congested/busy, so the next extension would be 102, if I wanted
VoiceMail to kick in in that case, this works:
exten = 12345,1,Dial(SIP/me,30)
exten = 12345,2,VoiceMail(su12345)
exten = 12345,102,VoiceMail(su12345)

But that is not, what I had in mind, I would like to have 30 seconds to
get to the phone, so in theory, this should do the trick:
exten = 12345,1,Dial(SIP/me,30)
exten = 12345,2,VoiceMail(su12345)
exten = 12345,102,Wait(30)
exten = 12345,103,VoiceMail(su12345)

But Asterisk can not take over the line after the wait.

To test, if the Wait was the problem, I created this:
exten = 12345,1,Wait(10)
exten = 12345,2,Answer()
exten = 12345,3,Milliwatt()

And still: Asterisk can't take over the ISDN line. The console output
says:
  == ISDN1: Incoming call '12345' - '12345'
-- Executing Wait(CAPI/ISDN1/12345-19, 10) in new stack
-- Executing Answer(CAPI/ISDN1/12345-19, ) in new stack
  == ISDN1: Answering for 12345
-- Executing Milliwatt(CAPI/ISDN1/12345-19, ) in new stack
CAPI INFO 0x34d1: Invalid call reference value
  == Spawn extension (capi-in, 12345, 3) exited non-zero on
'CAPI/ISDN1/12345-19'
  == ISDN1: CAPI Hangingup

If I try that in a pure sip-context, it works as I thought it would.

Now: do I do something wrong? Is there a problem with the Wait()
application? Or is that more likely a bug in chan_capi-cm?

Regards, Florian.
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RE: [Asterisk-Users] routing question: multipath routing for SIP

[gw]

Yes, and, you will probably need a different 
method.

Are these t1's to the same provider? Have you 
considered bonding the channels?

Greg


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Script 
HeadSent: Thursday, February 02, 2006 6:32 PMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: 
[Asterisk-Users] routing question: multipath routing for 
SIP
I have two T1s and I'd like to split my SIP traffic over the two. 
I am looking at this:http://lartc.org/howto/lartc.rpdb.multiple-links.htmlwhat 
bothers me about it is the note "Note that balancing will not be 
perfect, as it is route based, and routes are cached. This means that routes to 
often-used sites will always be over the same provider.". If all my traffic goes 
to the same IP, which is a remote SER proxy, will my second T1 be utilized at 
all? Does anyone have any experiece with this?
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RE: [Asterisk-Users] Re: delaying answer for a number of ringsor an amount of time

[gw]

No, it will dial like a pass-through simultaneously to sip/iax
extensions.  If you were to dial out to an analog port though, that
would be different.

So in essence, you can have all the phones ringing at the same time.

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian J.
Murrell
Sent: Thursday, February 02, 2006 7:46 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Re: delaying answer for a number of
ringsor an amount of time

On Thu, 2006-02-02 at 15:24 -0700, Bromont Quebec wrote:
 You need to take that Wait and Answer out of there
 
 [from-pots]
 exten = s,1,Dial(SIP/brianSIP/joe,30) exten = s,2,Voicemail(u2001) 
 exten = s,3,Hangup exten = s,102,Voicemail(b2001) exten = 
 s,103,Hangup exten = h,1,Hangup exten = i,1,Hangup

How does doing only that prevent Asterisk from picking up the POTS line
for a period of time (like 3 or 4 rings... or 10 seconds or so to give a
handset on the same POTS line an opportunity to pick it up first --
think answering machine)?  As I understand it removing the Wait and
Answer would cause Asterisk to pick the POTS line up right away and dial
brian and joe's phones with it.

Am I missing something?

b.

--
My other computer is your Microsoft Windows server.

Brian J. Murrell
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RE: Re: [Asterisk-Users] No IN and OUT on ISDN line at the same

[gw]

This could be a context issue, I had to fuss with mine to get the
channels working independently too.

I'll try to post the examples tomorrow, way to tired now :).

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Armin
Schindler
Sent: Saturday, January 28, 2006 4:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: Re: [Asterisk-Users] No IN and OUT on ISDN line at the same

On Sat, 28 Jan 2006, Ralf Mueller wrote:
 Hello Armin,
 
  The card is telling:
 CAPI INFO 0x34a2: No circuit / channel available
  
  so the other channel must be in use by something else.
  Maybe another device on the ISDN line?
  
 I have tested it several times now and always entered capi info
before and after the call.
 The answer was always:
 
 Contr1: 2 B channels total, 2 B channels free.

Okay, that means that Asterisk/chan_capi isn't using a channel at that
time. 
But it does not know about other programs or even other devices on the
ISDN bus.
When the call is coming in, are you sure you don't try to forward it to
more than one CAPI destinations? For each destination, one channel is
needed, even if the call is not accepted.
 
 I'm currently alone in the office, no incoming/outgoing faxes, no
incoming/outgoing calls.
 Is there a chance for me to figure out who or what is using the other
B channel while the call is coming in?

A dchannel trace might show something.

Armin
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RE: Re: [Asterisk-Users] No IN and OUT on ISDN line at the same

[gw]

Ok my examples are here for capi:

Simple but works.

http://www.voip-info.org/wiki/view/Example+North+American+CAPI+Setup

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gregory
Wiktor - ADCom Corp.
Sent: Saturday, January 28, 2006 4:53 AM
To: asterisk-users@lists.digium.com
Subject: RE: Re: [Asterisk-Users] No IN and OUT on ISDN line at the same

This could be a context issue, I had to fuss with mine to get the
channels working independently too.

I'll try to post the examples tomorrow, way to tired now :).

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Armin
Schindler
Sent: Saturday, January 28, 2006 4:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: Re: [Asterisk-Users] No IN and OUT on ISDN line at the same

On Sat, 28 Jan 2006, Ralf Mueller wrote:
 Hello Armin,
 
  The card is telling:
 CAPI INFO 0x34a2: No circuit / channel available
  
  so the other channel must be in use by something else.
  Maybe another device on the ISDN line?
  
 I have tested it several times now and always entered capi info
before and after the call.
 The answer was always:
 
 Contr1: 2 B channels total, 2 B channels free.

Okay, that means that Asterisk/chan_capi isn't using a channel at that
time. 
But it does not know about other programs or even other devices on the
ISDN bus.
When the call is coming in, are you sure you don't try to forward it to
more than one CAPI destinations? For each destination, one channel is
needed, even if the call is not accepted.
 
 I'm currently alone in the office, no incoming/outgoing faxes, no
incoming/outgoing calls.
 Is there a chance for me to figure out who or what is using the other
B channel while the call is coming in?

A dchannel trace might show something.

Armin
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RE: [Asterisk-Users] Re: Polycom 501 horrible echo

[gw]

Can someone post the sample files somewhere for 1.6.2?  I may have the
same issue but the firmware dl from voipsupply I believe did not include
the newer samples...

Thanks,
Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ron
Senykoff
Sent: Saturday, January 28, 2006 10:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Polycom 501 horrible echo

 One thing I was pondering: you are not, by chance, using the same 
 sip.cfg between version 1.4.1 and version 1.6.2 are you?  The file has

 changed significantly between these versions, and certain acoustic 
 settings that worked with 1.4.1 may not work with 1.6.2 (Not to 
 mention that ipmid.cfg and sip.cfg were merged in the 1.5.x release).

That has got to be the problem! I'll let you know how the results go.

Thanks
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RE: [Asterisk-Users] Chan_capi on builds 79558320 strangeness

[gw]

 /etc/init.d/asterisk stop
Stopping Asterisk PBX: .
censys:/usr/src/asterisk-8632#  cd ..
censys:/usr/src# asterisk -vc

  == Parsing '/etc/asterisk/asterisk.conf': Found

  == Parsing '/etc/asterisk/extconfig.conf': Found

Asterisk SVN-trunk-r8620, Copyright (C) 1999 - 2006 Digium, Inc. and
others.

Created by Mark Spencer [EMAIL PROTECTED]

Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for
details.

This is free software, with components licensed under the GNU General
Public

License version 2 and other licenses; you are welcome to redistribute it
under

certain conditions. Type 'show license' for details.


=

  == Parsing '/etc/asterisk/logger.conf': Found

Asterisk Event Logger Started /var/log/asterisk/event_log

Asterisk Dynamic Loader loading preload modules:

CLIP
 [chan_capi.so] = (Common ISDN API for Asterisk)

  == Parsing '/etc/asterisk/capi.conf': Found

  == This box has 1 capi controller(s).

-- CAPI/contr1 supports DTMF

-- CAPI/contr1 supports echo cancellation

-- CAPI/contr1 supports line interconnect

-- CAPI/contr1 supports supplementary services

supplementary services : 0x010f

HOLD/RETRIEVE

TERMINAL PORTABILITY

ECT

3PTY

MWI

  == Reading config for ISDNL1

-- capi_pvt ISDNL1-pseudo-D (5912211,capi-in-5912211,0,2) (1,4,64)

-- capi_pvt ISDNL1 (5912211,capi-in-5912211,0,2) (1,4,64)

-- capi_pvt ISDNL1 (5912211,capi-in-5912211,0,2) (1,4,64)

  == Reading config for ISDNL2

-- capi_pvt ISDNL2-pseudo-D (6930821,capi-in-6930821,0,2) (0,0,64)

-- capi_pvt ISDNL2 (6930821,capi-in-6930821,0,2) (0,0,64)

-- capi_pvt ISDNL2 (6930821,capi-in-6930821,0,2) (0,0,64)

-- listening on contr1 CIPmask = 0x1fff03ff

  == Registered channel type 'CAPI' (Common ISDN API Driver (cm-0.6.3) )

  == Registered application 'capiCommand'

  == Registered custom function VANITYNUMBER

CLIP

Asterisk Ready.
*CLI capi debug CAPI Debugging Enabled
*CLI -- Saved useragent
PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067 for peer 364

-- Executing Set(SIP/366-11b2,
CALLFILENAME=/var/spool/asterisk/monitor/outgoing/9145912211/Out-200601
18-030458-9145912211_ADCOM Office_19145912211) in new stack

-- Executing SetCallerID(SIP/366-11b2, 9145912211) in new stack

-- Executing Monitor(SIP/366-11b2,
wav|/var/spool/asterisk/monitor/outgoing/9145912211/Out-20060118-030458
-9145912211_ADCOM Office_19145912211) in new stack

-- Executing Dial(SIP/366-11b2, IAX2/ll/19145912211) in
new stack

-- Called ll/19145912211

-- Call accepted by 208.139.204.232 (format ulaw)

-- Format for call is ulaw

-- IAX2/teliaxcsi-8 is making progress passing it to SIP/366-11b2

-- Saved useragent Aastra 480i Cordless/1.3.0.1080 Brcm
Callctrl/1.5 MxSF/v3.2.6.26 for peer 347

-- Saved useragent Aastra 480i Cordless/1.3.0.1080 Brcm
Callctrl/1.5 MxSF/v3.2.6.26 for peer 345

-- Saved useragent Aastra 480i Cordless/1.3.0.1080 Brcm
Callctrl/1.5 MxSF/v3.2.6.26 for peer 361

-- Saved useragent Aastra 480i Cordless/1.3.0.1080 Brcm
Callctrl/1.5 MxSF/v3.2.6.26 for peer 363

-- Hungup 'IAX2/teliaxcsi-8'

  == Spawn extension (cisco-teliaxoutcsi, 19145912211, 4) exited
non-zero on 'SIP/366-11b2'

-- Saved useragent PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067 for
peer 365

-- Saved useragent PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067 for
peer 330

-- Executing Set(SIP/366-5e8d,
CALLFILENAME=/var/spool/asterisk/monitor/outgoing/9145912211/Out-200601
18-030510-9145912211_ADCOM Office_19145912211) in new stack

-- Executing SetCallerID(SIP/366-5e8d, 9145912211) in new stack

-- Executing Monitor(SIP/366-5e8d,
wav|/var/spool/asterisk/monitor/outgoing/9145912211/Out-20060118-030510
-9145912211_ADCOM Office_19145912211) in new stack

-- Executing Dial(SIP/366-5e8d,
IAX2/[EMAIL PROTECTED]/19145912211) in new stack

-- Called [EMAIL PROTECTED]/19145912211

-- Call accepted by 208.139.204.232 (format ulaw)

-- Format for call is ulaw
Jan 17 22:05:11 WARNING[8571]: chan_iax2.c:7525 socket_read: Received
mini frame before first full voice frame
 
-- IAX2/teliaxcsi-9 is making progress passing it to SIP/366-5e8d

CONNECT_IND ID=001 #0x0001 LEN=0050
  Controller/PLCI/NCCI= 0x201
  CIPValue= 0x1
  CalledPartyNumber   = c15912211
  CallingPartyNumber  = 21 819145912211
  CalledPartySubaddress   = default
  CallingPartySubaddress  = default
  BC  = 80 90 a2
  LLC = default
  HLC = default
  AdditionalInfo 
   BChannelinformation= default
   Keypadfacility = default
   Useruserdata   = default
   Facilitydataarray  = default


-- CONNECT_IND

RE: [Asterisk-Users] Chan_capi on builds 79558320 strangeness

[gw]

Strange though it's only effecting since build 8000...

Here's the snippet:

exten = s,1,LookupCIDName
exten =
s,2,Set(CALLFILENAME=/var/spool/asterisk/monitor/incoming/${IncomingLine
}/In-${STRFTIME(${EPOCH},,%Y%m% . . .
exten = s,3,Monitor(wav,${CALLFILENAME})
exten = s,4,GotoIf($[${INCOMINGLINE} = 9146930821]?9:5);
exten = s,5,GotoIfTime(20:01-7:59|mon-sun|*|*?9)
exten = s,6,Dial(${ADCOMDAYRINGTO},25,t);All
exten = s,7,NoOp(${DIALSTATUS})
exten = s,8,Goto(adcomincoming,s,11)
exten = s,9,Dial(${ADCOMNIGHTRINGTO},25,t);Cisco,Ping,Poly,SPA841
exten = s,10,NoOp(${DIALSTATUS})  
exten = s,107,Answer
exten = s,108,Wait(1)
exten = s,109,BackGround(adcom1/thankyou); Thank you for calling ADCOM
Corp.
exten = s,110,Playback(busy-pls-hold)
exten = s,111,Queue(adcomgwqueue)
exten = s,11,Answer ; Answer the line
exten = s,12,Wait(1)

... Menu plays.

ADCOMDAYRINGTO =
${C79601L1}${OFFICE3}${POLY1L1}SIP/344SIP/345SIP/364${SOMERSADCOM}
; SIP/355SIP/342 SP
ADCOMNIGHTRINGTO =
${C79601L1}${POLY1L1}SIP/344SIP/345SIP/364${SOMERSADCOM} 

So could it have something to do with the dialstring?  I would think
asterisk would say something first before doing the dials.

I'll try it later with a simple dialstring.  I'm going to rebuild it
anyhow.

I am looking to use a global variable in like a switch setup, to direct
calls to particular setups based on a menu.  For example, someone dials
ext 333, and they get a menu for day mode, night mode, holiday, away
from office, etc and the dialplan will ring different devices depending
on the choice...

We'll see what happens...

On a side note, I believe it works if I dial right into the menu
playback.  But if it's the dialstring that's wrong, I would think
asterisk should complain about it.

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Armin
Schindler
Sent: Friday, January 27, 2006 4:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Chan_capi on builds 79558320 strangeness

This is not a problem of the ISDN line (or chan_capi), Asterisk is just
not doing anything after

  -- Executing
GotoIfTime(CAPI/ISDNL1/5912211-0,20:01-7:59|mon-sun|*|*?9) in new
stack

and without further commands (like Ringing(), Answer(), ...) the ISDN
line timed out and disconnects.

So either your dialplan is buggy, or Asterisk is not doing what you
want.
What should be done according your extensions.conf in that state ?

Armin

On Fri, 27 Jan 2006 [EMAIL PROTECTED] wrote:
  /etc/init.d/asterisk stop
 Stopping Asterisk PBX: .
 censys:/usr/src/asterisk-8632#  cd ..
 censys:/usr/src# asterisk -vc
 
   == Parsing '/etc/asterisk/asterisk.conf': Found
 
   == Parsing '/etc/asterisk/extconfig.conf': Found
 
 Asterisk SVN-trunk-r8620, Copyright (C) 1999 - 2006 Digium, Inc. and 
 others.
 
 Created by Mark Spencer [EMAIL PROTECTED]
 
 Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for 
 details.
 
 This is free software, with components licensed under the GNU General 
 Public
 
 License version 2 and other licenses; you are welcome to redistribute 
 it under
 
 certain conditions. Type 'show license' for details.
 
 ==
 ==
 =
 
   == Parsing '/etc/asterisk/logger.conf': Found
 
 Asterisk Event Logger Started /var/log/asterisk/event_log
 
 Asterisk Dynamic Loader loading preload modules:
 
 CLIP
  [chan_capi.so] = (Common ISDN API for Asterisk)
 
   == Parsing '/etc/asterisk/capi.conf': Found
 
   == This box has 1 capi controller(s).
 
 -- CAPI/contr1 supports DTMF
 
 -- CAPI/contr1 supports echo cancellation
 
 -- CAPI/contr1 supports line interconnect
 
 -- CAPI/contr1 supports supplementary services
 
 supplementary services : 0x010f
 
 HOLD/RETRIEVE
 
 TERMINAL PORTABILITY
 
 ECT
 
 3PTY
 
 MWI
 
   == Reading config for ISDNL1
 
 -- capi_pvt ISDNL1-pseudo-D (5912211,capi-in-5912211,0,2) (1,4,64)
 
 -- capi_pvt ISDNL1 (5912211,capi-in-5912211,0,2) (1,4,64)
 
 -- capi_pvt ISDNL1 (5912211,capi-in-5912211,0,2) (1,4,64)
 
   == Reading config for ISDNL2
 
 -- capi_pvt ISDNL2-pseudo-D (6930821,capi-in-6930821,0,2) (0,0,64)
 
 -- capi_pvt ISDNL2 (6930821,capi-in-6930821,0,2) (0,0,64)
 
 -- capi_pvt ISDNL2 (6930821,capi-in-6930821,0,2) (0,0,64)
 
 -- listening on contr1 CIPmask = 0x1fff03ff
 
   == Registered channel type 'CAPI' (Common ISDN API Driver (cm-0.6.3)

 )
 
   == Registered application 'capiCommand'
 
   == Registered custom function VANITYNUMBER
 
 CLIP
 
 Asterisk Ready.
 *CLI capi debug CAPI Debugging Enabled
 *CLI -- Saved useragent
 PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067 for peer 364
 
 -- Executing Set(SIP/366-11b2,
 CALLFILENAME=/var/spool/asterisk/monitor/outgoing/9145912211/Out-2006
 01 18-030458-9145912211_ADCOM Office_19145912211) in new stack
 
 -- Executing 

RE: [Asterisk-Users] Polycom 501 horrible echo

[gw]

Hello Chad,
Where did you get 1.6.4.0064?  Site says latest is 1.63.0067.  Also my
supplier only has 1.63.0067.

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chad
Osmond
Sent: Friday, January 27, 2006 11:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Polycom 501 horrible echo

I've been running 1.6.4.0064 for the last few weeks..
I've had no problems with it, I haven't done a whole lot of speaker
phone with it yet though.. Once my IP4000 reboots It'll be running it as
well so that will be a good test.

Chad 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff
Herring
Sent: January 26, 2006 7:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom 501 horrible echo

Now I'm really confused...
1.6.3 is on the Polycom Website as the latest...

I'm running 1.6.2.0041 according to my phone.

Which firmware worked for you?

At 04:04 PM 1/26/2006, Ron Senykoff wrote:
  We also have noticed a poor server config can cause this in testing.
 
  Noticed when I had one person building * servers using Debian. Had 
  them rebuilt with FC4 and have no issues - yet:)

I recently upgraded all our phones to the latest Polycom firmware
1.6.2 and went from great speakerphone to tons of feedback. I would 
hate to have to go back to the old firmware. Although Polycom 
recommends keeping the older bootrom unless you need https 
provisioning, I'm going to try the new bootrom and see if it fixes the 
problem.

This is being experienced across 3 corporate offices with 3 separate 
Asterisk servers. And I have to reiterate... all was good until the 
firmware upgrade.

-Ron
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[Asterisk-Users] Chan_capi on builds 79558320 strangeness

[gw]

Hello All,
I am having an odd problem with Armin's chan-capi_cm on builds higher
than 7955.

It would seem that this happens on anything higher than 7955.

What is happening is the isdn is ringing, then asterisk does a goto-if
and just hangs.  

Asterisk itself is ok, but the isdn then rings out or busys out on the
other side.

Outgoing works fine, this only seems to effect incoming.

I updated to chan-capi_cm 0.6.3 but there is no change.

Noticed this when trying to update for the timebomb bug.

I think it is somehow related to the dial command but I'm not certain.

Has anyone else experienced such oddness?

Regards,
Greg
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RE: [Asterisk-Users] Linksys SPA-941 multiple line appearences

[gw]

Did you by any chance try it with call waiting enabled on the spa?

I think that's how I have one of mine and it works fine.


Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Keyes
Sent: Wednesday, January 25, 2006 10:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Linksys SPA-941 multiple line appearences

I set up only 1 extension and set all 4 line appearences to point to
that extension.  I could place up to 4 outgoing calls as extension 1 no
problem.
The problem happened when there was an active call on line appearence 1
and someone called extension 1.  Instead of ringing on line appearence 2
(extension 1) it would busy out.  My question, is there something in
Asterisk that needs to be adjusted to make the phone work properly or is
there a setting on the phone that needs to be adjusted?  Thank you.

Michael K


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kerry
Garrison
Sent: Tuesday, January 24, 2006 10:33 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Linksys SPA-941 multiple line appearences


You only need to setup ONE account and all four call appearances will
work.
-Kerry


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Michael 
 Keyes
 Sent: Tuesday, January 24, 2006 3:41 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Linksys SPA-941 multiple line appearences

 Has anyone had any experience with the Linksys SPA-941 when it comes 
 to multiple line appearences?

 The 941 comes with 4 line appearence buttons which can individualy be 
 configured to point at any extension.  The phone is capable of 2 
 extensions out of the box with the option to add  2 more for a license

 fee.

 The 941 manual states that, The Call Waiting function is activated 
 when a device has a call in the active state and another call is 
 incoming. The phones in the SPA series do not support multiple calls 
 on the same Line Key.
 Incoming calls are assigned to an unused Line Key, causing the Line 
 Key to quickly blink red. (Note that the Voice Mail Waiting Indicator 
 also blinks red whenever there is an incoming call.) The phone will 
 not ring. However, to alert the user, the call waiting tone is played 
 into the active audio device.

 During testing I set up an Asterisk 1.2 box with a 941 phone using 
 firmware ver. 4.1.8.  I configured 1 extension and set all 4 line 
 appearence buttons to point to that extension.  If there was an active

 call in progress I could place that call on hold and by pressing line 
 appearence button 2 was able to place an outgoing call.  That outgoing

 call would appear to come from extension 1.  This is all working as 
 desired.

 If an active call was in progress and someone called my extension the 
 product manual indicates that call should appear on line appearence 
 button 2.  During my testing Asterisk would flag my extension 1 as 
 busy and instead of ringing the phone on line appearence 2 would send 
 the call to voicemail.

 Is anyone aware of any configuration setting needed on either the 
 Asterisk server or on the phone to make Call Waiting function as 
 described in the manual?  Thank you.

 Michael K


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[Asterisk-Users] Anyone using verizon fios ftth for analog voice? Any echo?

[gw]

Hello All,
I was wondering, is anyone using verizon's fios going into a zaptel 4
port card?

If so, has anyone experienced echo issues at all?

I am under the assumption that echo issues should be minimal on a ftth
connection, but want to confirm if this is the case.

I have some customers with nasty echo and fios is an option, but if it's
not likely to solve anything, would end up just being a lot of trouble.

Thanks,
Greg
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[Asterisk-Users] MOH begin behavior

[gw]

Hello All,
Does anyone know if you can start an MOH queue on an individual call?
What I mean is, for example if you have a script that you want the moh
to start with certain phrases, can it be done, or are you limited to the
standard looping audio?

It's almost like starting a stream for each call, and terminating it
when the call comes off of hold.

Regards,
Greg
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RE: [Asterisk-Users] Re: Anyone using verizon fios ftth foranalogvoice?Any echo?

[gw]

Thanks for answering my question guys :)

Guess I'll have to try it and find out, but will keep everyone posted...

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Tuesday, January 24, 2006 7:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Re: Anyone using verizon fios ftth
foranalogvoice?Any echo?



 -Original Message-
 From: JP Carballo [mailto:[EMAIL PROTECTED]
 Sent: Tuesday, January 24, 2006 7:32 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Re: Anyone using verizon fios ftth for 
 analogvoice?Any echo?
 
 Chris Mason (Lists) wrote:
 
  JP Carballo wrote:
 
  I would if I could. I'm paying more to Time Warner for a fraction
of
  the speed Verizon provides.
  Until Verizon is done with their tedious wiring of affluent 
  neighborhoods, I can only dream.
 
  Rich bastards. Come the revolution...
 
 
 lol!
 
 It's all relative.
 I know of an island in Asia where the people have to ride a ferry for
an
 hour  to the next island  so they can call their relatives here in the

 USA.
 The internet cafe they use is paying the same amount in dollars as a 
 typical yahoo dsl client for a line that's around a third of the
speed.
 
 I'm pretty sure they're saying the same about us Chris. :)
 
 --
 JP Carballo
 

How about paying over $3,000 U$D a month for a voice E1 in Senegal!
VSAT is almost cheaper.

Thanks,
Steve
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RE: [Asterisk-Users] Polycom 501 horrible echo

[gw]

I have had the same issue.  It has a lot to do with the acoustics, as
well as gain.  Before I messed with the config files it sounded great,
then I fussed with them and upgraded to the latest sip, and now I also
notice this on speaker. 

I would go totally default, local configure and see how they sound...

Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Monday, January 23, 2006 9:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Polycom 501 horrible echo

You aren't making calls from one phone to another, with them right next
to each other on the same desk are you? 
 
Doug.

-Original Message- 
From: Jeff Herring [mailto:[EMAIL PROTECTED] 
Sent: Mon 1/23/2006 6:46 PM 
To: asterisk-users@lists.digium.com 
Cc: 
Subject: [Asterisk-Users] Polycom 501 horrible echo



I have the following situation:

Asterisk 1.2.1
25+ Polycom 501 telephones. Bootrom 2.6.2.0032 Application
1.6.2.0041
Some 501's local to my network, some across the great INTERNET
divide.
PRI connected to Sangoma card.

Issue: horrible echo (and squeals, and underwater-like sound)
on speaker
phone when calling from extension to extension.

echo not present when calling outbound using PRI or when
receiving calls
from PRI.
echo not present when using handset or headset in any case.

All gains, etc. are as listed in the Polycom Admin Guide.

Not specific to any phone, or its location on our network.

I suspect the issue is related to the echo cancelation HW in the
speaker
phone, but
I'm not sure...The unfortunate thing is these phones were
purchased because
of their
excellent speaker phones which now appear to be worse than the
Grandstreams!

Anyone with thoughts of where to start?

TIA - Jeff H.

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RE: [Asterisk-Users] Hardwiring a Tellabs echo canceller - help req

[gw]

Dan, Basically you go send tip  send tip, receive tip  receive tip and
so forth on both connectors.

I needed to use one cat 5 out of the ec to the channel bank, and one
crossover t1 cable to the zaptel card.  Receive in means receive
basically (because it's the incoming signal).

You should be able to loop the card to see if it's working, either with
a straighthrough or a crossover, I don't recall If it loops you'll
get AIS Loop on the card (you'll notice it)

Greg



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Elder
Sent: Friday, January 20, 2006 3:14 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Hardwiring a Tellabs echo canceller - help req

Hi All, Greg has been a huge help getting me going with this tellabs
echo can, but I'm still having some problems getting it to work... I
suspect I wired it up incorrectly, so I thought I'd see if anyone can
point me in the right direction. Digium tech support pointed me to this
doc for a standard
T1 cable:

http://www.arcelect.com/RJ48C_and_RJ48S_8_position_jack_.htm

which looks like it means:

pin1 = recieve ring
pin2 = recive tip
pin4 = send ring
pin5 = send tip


So I treated the bottom part of the diagram like a noraml cat5 cable 
connected it in this way:

Zap side | 2572 side

pin 1 = pin 25
pin 2 = pin 21
pin 4 = pin 10
pin 5 = pin 6


on the T1 side I have:

T1 side | 2572 side
-
pin1  = 26
pin2  = 22
pin4  = 9
pin5  = 5

In the wiki page for these pinouts are listed as
T1 side(drop)

  5 Send In Tip
  9 Send In Ring
 22 Receive Out Tip
 26 Receive Out Ring

Zap side (line)

 21 Receive In Tip
 25 Receive In Ring
  6  Send Out Tip
 10 Send Out Ring

and I'm not sure what the 'in'  'out' desinations mean

can anyone help me or point me in the right direction for wiring this
bugger up? perhaps I did do it correctly  the can isn't configured yet
(just did that, but can't test till 2night when the office clears out)

Thanks in advance for any insights

Dan

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[Asterisk-Users] Tellabs 2572 EC Photos here.

[gw]

Hello Dan,

Have a look at this link:

http://www.adcomcorp.com/asterisk/tellabs

I got those pictures up there, may be of help.  In essence, 1 pair is
either a tx pair or an rx pair.

If I recall, Orange and Green should be one side, and blue brown should
be the other side.

I tried to upload them to the wiki (even 40% size) but they didn't show
up for some reason...

Files are large on the page...

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Elder
Sent: Friday, January 20, 2006 3:14 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Hardwiring a Tellabs echo canceller - help req

Hi All, Greg has been a huge help getting me going with this tellabs
echo can, but I'm still having some problems getting it to work... I
suspect I wired it up incorrectly, so I thought I'd see if anyone can
point me in the right direction. Digium tech support pointed me to this
doc for a standard
T1 cable:

http://www.arcelect.com/RJ48C_and_RJ48S_8_position_jack_.htm

which looks like it means:

pin1 = recieve ring
pin2 = recive tip
pin4 = send ring
pin5 = send tip


So I treated the bottom part of the diagram like a noraml cat5 cable 
connected it in this way:

Zap side | 2572 side

pin 1 = pin 25
pin 2 = pin 21
pin 4 = pin 10
pin 5 = pin 6


on the T1 side I have:

T1 side | 2572 side
-
pin1  = 26
pin2  = 22
pin4  = 9
pin5  = 5

In the wiki page for these pinouts are listed as
T1 side(drop)

  5 Send In Tip
  9 Send In Ring
 22 Receive Out Tip
 26 Receive Out Ring

Zap side (line)

 21 Receive In Tip
 25 Receive In Ring
  6  Send Out Tip
 10 Send Out Ring

and I'm not sure what the 'in'  'out' desinations mean

can anyone help me or point me in the right direction for wiring this
bugger up? perhaps I did do it correctly  the can isn't configured yet
(just did that, but can't test till 2night when the office clears out)

Thanks in advance for any insights

Dan

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[Asterisk-Users] Includes affecting menu, zaptel transfers

[gw]

Hello all,

I have a minor annoyance and I think there is a better solution...

I have a dialin menu, with options 1 and 2.  

I need the ability to do a zaptel flash transfer, but to get the proper
extensions to work, I need to do an include of the context which
contains those extensions.  Is there a way to include the context, but
have the menu override so there is not a 5 second delay after the user
presses the key from the s menu?

Any thoughts?

Greg
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RE: [Asterisk-Users] Adit 600 and echo

[gw]

 

-Original Message-
From: Gregory Wiktor - ADCom Corp. 
Sent: Sunday, January 15, 2006 3:29 PM
To: 'Patrick'
Subject: RE: [Fwd: RE: [Asterisk-Users] Adit 600 and echo]

Hello Patrick,
I believe mine is the 2572.  64ms EC.

Power supply:
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=7574802048

Yes got them off ebay, but keep your eye open.  I just checked, I got
the EC for only 30.49 including shipping from ebay :)

http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5824630277

Yay...

I'll try to take some pictures this week for the wiki...

The soldering is easy but it will require good quiet time.  Then I ran
the lines 2 a 2 port cat6 block.  

Greg

-Original Message-
From: Patrick [mailto:[EMAIL PROTECTED]
Sent: Sunday, January 15, 2006 10:22 AM
To: Gregory Wiktor - ADCom Corp.
Subject: [Fwd: RE: [Asterisk-Users] Adit 600 and echo]

 Forwarded Message 
 From: [EMAIL PROTECTED]
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 To: asterisk-users@lists.digium.com
 Subject: RE: [Asterisk-Users] Adit 600 and echo
 Date: Sun, 15 Jan 2006 03:50:10 -0500
 
 See my other post.  I had the exact same problem on about 5 lines. 
 
 I got a tellabs ec a while back, and then a power supply but hadn't 
 the time to solder it up and get things going...I had just been 
 forwarding the inbound calls to voicepulse connect.
 
 I finally did get it soldered and ready, and when I installed the new 
 hardware tellabs ec, echo was gone.  I must have spent 10-20 hours on 
 this problem, and the tellabs solved it in 5 minutes (once setting the

 channel properties to FXO-LS  FXS-LS)
 
 I think it cost something like $110 for the card, and $86 for the 
 power supply.  Then you need the time to do the soldering.

Greg,

Can you please tell me the modelnumber of the Tellabs' card and power
supply and any info/links regarding the soldering? Where did you buy the
card and power supply, eBay?

Thanks and regards,
Patrick
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[Asterisk-Users] Oooh / ahhh . . . 5 tellabs boards on ebay.

[gw]

http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5854671489ssPageName
=MERC_VIC_ReBay_Pr4_PcY_BIN_Stores_IT#ebayphotohosting

Worth considering for some . . . :)

I got my unit from the same fellow, worked out fine...

Greg
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RE: [Asterisk-Users] SIP phones unbeatable echo

[gw]

Hello Dan,

I was fighting with echo on a number of circumstances, and came to the
following conclusions.

If you are on a distant loop, or analog lines with issues, those issues
need to be addressed or you need a workaround.

In a few cases, I converted to ISDN-BRI, which has been one of my best
decisions, because I get excellent quality as well as high-speed call
completion...

In one case, I put in an ADIT 600 channel bank, and still had serious
echo problems. I tried and tried, but found no simple solution by
messing with the zapata drivers.  Installing a hardware tellabs echo
canceller totally solved the echo issue.  I have the zapata.conf echo
cancellation totally off, and the lines sound great.  These are also
lines that are odd, meaning about 15K feet from the CO, with periodic
instabilities during rain/snow.

I went through the various tweaks, milliwatt tests, etc, but only the
hardware could solve it (and in minutes after installation as opposed to
the hours I spend working with software).

Depending on the amount of channels you have, you may consider a
channelbank with tellabs, or one of the new digium analog cards with ec,
though I have not used the new digiums yet myself.  They are expensive
solutions, but the best solutions too.

I wish there were 4 port card that had great EC, but there isn't.  I
wait for the day that we have pci-express voip cards at our disposal,
that would be something...  Asterisk would take off entirely at that
point, since the latencies that cause so many problems would be gone,
and the capacities would be so much higher.

Just in case I went over your head here, sipsip should produce no echo.
If it does there are other issues.  If you are going analoganalog and
hear no echo, I would have a look at the network itself.

Regards,
Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Elder
Sent: Thursday, January 12, 2006 2:53 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP phones unbeatable echo

Hey all again, I'm wrestling with echo problems on our sip extensions.
I've set these items in zapata.conf but tweaking these values doesn't
seem to make much difference


echocancel=yes
echocancelwhenbridged=yes
echotraining=2500
rxgain=8.0
txgain=1.0


are there other settings that can help me tame this beast? Been
searching but not turning up anything that'll work here.

Thanks in advance.

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RE: [Asterisk-Users] Asterisk crossed lines?

[gw]

 Could this be a situation where asterisk is picking up a line just as
the call comes in?

E.G. it's common when an incoming call comes in, someone goes to dialout
and answers instead, listens, and in most cases just hangs up or picks a
new line...

Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan Elder
Sent: Thursday, January 12, 2006 12:03 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk crossed lines?

Hey all, been noticing some oddness on a new AAH install... occasionally
an incoming zap line with automatically connect with an outgoing
extension, even though the incoming line hasn't specified what extension
it's aiming for (i.e. haven't tapped in the ext # yet)... so someone's
trying to call out from inside the office  are automatically connected
with an incoming line. Anyone seen this or know what might be causing
it?

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RE: [Asterisk-Users] Adit 600 and echo

[gw]

See my other post.  I had the exact same problem on about 5 lines. 

I got a tellabs ec a while back, and then a power supply but hadn't the
time to solder it up and get things going...I had just been forwarding
the inbound calls to voicepulse connect.

I finally did get it soldered and ready, and when I installed the new
hardware tellabs ec, echo was gone.  I must have spent 10-20 hours on
this problem, and the tellabs solved it in 5 minutes (once setting the
channel properties to FXO-LS  FXS-LS)

I think it cost something like $110 for the card, and $86 for the power
supply.  Then you need the time to do the soldering.  

In my case, being far from the CO, I found no other solution than the
hardware EC.  You do need some soldering experience though, because you
don't want to lift pads off the board(something I had fun with dealing
with SMD's and such).

I can say though, that the resulting quality of the line is so decent, I
am highly considering using the POTS for outgoing instead of iax
connections to ITSP's (because I don't like latency).

It really is an impressive difference.

If you have 13 lines, I am surprised you are using a channel bank (if
they are pots lines).  Wouldn't it be more cost effective to use a T1
from your CO?

Ztmonitor or a good tool, but in my experience just not enough when you
are on a distant loop.  I went through milliwatt tests like crazy trying
to tweak gains, with little luck.

In the end, the $200 or so I spent on hardware was totally worth it...

If you are not interested in doing the soldering, you can probably find
someone on the list who can supply you with a working EC setup

You're welcome to call me if you want to discuss what I went through...

Office: 1-914-591-2211
Mobile: 1-914-582-9110

Regards,
Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Thursday, January 12, 2006 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Adit 600 and echo

I'm having issues with echo.

My setup is Polycom IP501 phones connected to an Adit 600 via T100P.  13
Lines going to the Adit.

All echo so far is on the local side (Employees hears own voice, but
only on some calls).

Watching the channel with ztmonitor, I notice that TXGAIN is pegged out
most of the time.  RXGAIN is anywhere between 45 and 60%.

Is it safe to assume that playing around with the TX/RX gains on the
channel bank will not do anything and this needs to be resolved via the
TX gains within Asterisk?

I was able to set the TXGAIN to -6.3 and it did help some, but if I try
-6.4 or more on any of the channels, I can no longer here audio when
calling into the facility. 

Any suggestions?

Doug

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[Asterisk-Users] Tellabs echo can, can someone wire mine up for $?

[gw]

Hello All,
I have a tellabs 2572 EC, and I was wondering if anyone with experience
can wire this up for me...

I have the proper tools and info, but not the time.  It's easier to send
it out and get it done.  I know kb1_kanobe originally started the wiki
thread, and rather than risking the card, would it be possible for me to
send it out to him, have him do it, and send it back?

Not to say I am not capable, I have some nice re-work equipment, but 2
hours to me is worth $250.  I also prefer someone with the experience to
do it properly, rather than fussing with it. 

Or, does someone know where I can find a cabinet for a fare amount?
Although it's overkill for just one card.

I'd pay him, just too busy to deal with it right now... Or at least till
Feb :)  Mostly because my google adwords and yahoo advertising is
booming in my local area for me, tons of leads (I am in NY, doing
computer networking services).

Regards,
Greg
www.adcomcorp.com
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[Asterisk-Users] How does the PCI bus effect latency and echo?

[gw]

Hello All,
I was wondering, how does the PCI bus effect echo and latency?

I know a large part of the echo issues out there have to do with the pci
bus latency, but are there any suggestions on the hardware side to
minimize this?  For example, a 533mhz vs 800mhz fsb, would it have any
effect on communications?  Even on a low load server, I am trying to
decide if getting a dell poweredge server is worth the effect, versus
using an intel board or asus board.

Regards,
Greg
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RE: [Asterisk-Users] Wake-Up Call

[gw]

Something to think about is this too, when completed scheduling, ask
would you like to notify another extension, so if the first does not
answer in two attempts, ring a cell phone or such. 

But I cannot complain, I use the wakeup call function every day, and it
is definitely better than any alarm clock or pbx reminder available.

Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of trixter
aka Bret McDanel
Sent: Monday, January 09, 2006 6:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Wake-Up Call

On Mon, 2006-01-09 at 12:07 +0100, Tomislav Parcina wrote:
 I have setup wake up call in * following those instructions 
 http://www.voip-info.org/wiki/view/Asterisk+tips+Wake-Up+Call+PHP
 and it works fine. Now I have few questions. 
 
 - When I arrange wake up call, does it call me only that day or I 
 can set it up for whoole week?
 - Can I set it up for some other extension or only for one I'm
calling?
 - Can this AM, PM be in 24h format?
 
 That is all (for now :)).
 
 

That particular script appears to only schedule for the next 24 hours.
It could do more but it doesnt.  I was going to write a php one that
doesnt work quite this way.  Instead I was going to take advantage of
features of the queue app so you dont need a seperate cron job, namely
setting the time of the queue file to the time you want the wake up
call.  I was also going to add in features to record a custom message,
and other such goodies.  They arent complex features, but I think would
make it nicer.

But this is low priority for me right now.  Between the Sac AUG and ETEL
speaking engagements this month along with regular work I am unsure that
I will have time until feburary.

I hadnt thought about recurring ones, that would be better handled via a
crontab type setup I think than creating a ton of queue files.  You
could easily do this, just a matter of storing who, when, how
frequently, and then creating the queue files on time.  

If you are using the one I think you are then if you enter it in 24 hour
format and the time is  12 it can tell that you mean 24 hour format,
hwoever it cant tell teh difference bewteen 12hr and 24hr if the time is
 12 so it asks.


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group
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RE: [Asterisk-Users] Asterisk Jobs

[gw]

You make it sound so easy.  Even one customer that you sell, you may
make $1000 but your initial overhead is high.  You need to be full time
first of all, and also be capable of supporting your customers.

Just my 2 cents follows:

I have been in the IT business for over 15 years, and I cannot count how
many customers I have gotten because their previous consultants were
either incapable, dishonest, or unable to properly support them.

My personal opinion is that lack of support and capability in the IT
world is a certain problem.  I literally work 24/7, but I would not have
it any other way(at least until retirement).  

The voip market especially has problems with customer retention,
reliability, tech support, etc...  I feel this has been one of the
markets biggest problems.  It almost seems to make sense that voip
companies be licensed in some form to guarantee a level of reliability
and service that customers deserve.

Look at the options, there are some companies that are bad, and some
that are very good, but from the customers perspective no way to
distinguish between them except the feedback on lists and forums, or
from people they trust.  

For example, a number of my clients use Cablevision/Optimum online for
voice, and horridly hate the customer service and repair service.
Whereas others use verizon and really cannot complain nearly as much(not
to say they do not have problems, but verizon has gotten much better
recently).  It has gotten to the point where I suggest people not to use
optimum online, mostly because of their voice network, and the fact that
the are subbing most of their service personel.  Their internet works
well though. 

I am not promoting verizon in any way, sure they are inflated, but when
it comes to the operation of someones business or the effect of
technology on their personal life, they would simply rather not deal
with the complicated.  If the VOIP market is to make a really big boom,
we need an enterprise approach, mostly to customer service...

This is one reason why vonage has done well, they have taken an
enterprise approach to serving clients.  Sure they limit their
capability a lot (e.g. no iax/sip, etc) but they have a good business
model and it will no doubt serve them well in the future.

Ok, my 2 cents are over :)

Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Kalcevich
Sent: Saturday, January 07, 2006 9:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Jobs

I think it would be biggest is in consulting. The people that refuse or
cant  to pay for call manager or Avaya's one. Example asterisk 
sugarcrm.com they work together. Thats really good to sell. They arent
in monster.ca they are banging on doors making $.

Make a buch of pre setup asterisk configs that would be most popular
make marketing material, dump on website. go in trade shows. Demo and
make $


Steve kalcevich

Douglas Garstang wrote:

I'm curious why the number of jobs out there requiring Asterisk seems
to be pretty low. After looking around dice, monster, careerbuilder etc,
I was surprised to find no more than 3-4 employment opportunities with
Asterisk throughout the US.
 
Is it really that low? There seems to be a job of opportunities for
Cisco and other vendors solutions (duh... GUI's are good... duh). I
wonder if demand will increase, or am I just looking in the wrong
places?
 
- Doug.
 
  

---
-

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RE: [Asterisk-Users] Second edition of my * book has been released

[gw]

How does it compare with the O'Rielly book?

Does it include information on CVS, or primarily on stable?

Can it be provided to customers, or is it more sysadmin oriented?

Regards,
Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Mahler
Sent: Thursday, January 05, 2006 9:45 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Second edition of my * book has been released

The second edition of my Asterisk book VoIP Telephony with Asterisk is
now in print. It's reorganized and expanded. 

TKS

Paul Mahler


Paul Mahler
[EMAIL PROTECTED]
 
www.signate.com
   

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RE: [Asterisk-Users] Decent sub-$100 SIP phone.

[gw]

Get him a 940 sipura/linksys, big difference, and I can 
personally say they are 200% better.

I have used both and there is no comparison, from the 
user's end, it makes a big difference. One of the most important aspects 
ofa customers experience is the ease of use. If someone simply wants 
to make cheap calls, they do not care, but if they are looking for a good phone 
for a good price, the sipura 94x series are great. I personally prefer 
polycom/cisco. Polycom is great for multiple call handling, whereas the 
cisco 7960 is great for a remote worker.

Greg


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid B. 
Asterisk UsersSent: Tuesday, January 10, 2006 12:03 AMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] "Decent" 
sub-$100 SIP phone.

Ken,
I would tell the client that you offerd phones for 
under $100.00 and he didnt like them so now for a diffrent phone he will have to 
pay more. Also I have an 841 and for it works great. I also installed one for a 
customer in a mechanic shop and no complaints.

Regards,
Dovid
Message: 
15Date: Mon, 09 Jan 2006 15:28:28 -0500From: Ken D'Ambrosio 
[EMAIL PROTECTED]Subject: [Asterisk-Users] "Decent" sub-$100 SIP phone.To: 
Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comMessage-ID: [EMAIL PROTECTED]Content-Type: text/plain; charset=ISO-8859-1Hey, 
all. I quoted a customer about $100 for some cheap SIP phones. 
Iwas planning on using the BT-102's, but he called said they look 
like"Princess phones," and I have to admit that he has a point. Some 
of theother inexpensive phones look decent, but (for example) the 
SPA-841'swiki entry says the remote end gets a lot of static. Since 
it'll bebeing used from a noisy environment (a cleanroom), the less 
overallstatic, the better. Someone suggested the Polycom 301's, but 
I'd losemoney on them. [I'll go with them if I have to, as I'm making 
moneyelswhere, but still...] So, does anyone have any suggestions for 
decentsub-$100, professional-looking SIP phones?Thanks!Ken 
D'Ambrosio

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RE: [Asterisk-Users] Having major issues with TDM2400

[gw]

Hello Kerry,

Maybe it's me, but why are you using hint in this fashion?  Shouldn't
you be doing exten = 100,1,Dial(SIP/900zap/g0/w5551212) or is there
something new that I have missed?

Regards,
Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kerry
Garrison
Sent: Saturday, December 31, 2005 11:38 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Having major issues with TDM2400

To summarize, I spent 6 hours yesterday on the phone with Digium trying
to fix a problem with the TDM2400 ad we still don't have it working
right. The lastest version of everything are installed and confirmed by
Digium. So here is the issue:

Zapata.conf
; Disable call progress
; callprogress=yes

Outbound calls to PSTN phone numbers work properly

But using this:

exten = 100,hint,SIP/900zap/g0/w5551212

The extension will ring once, but as soon as the PSTN line is picked up,
the sip phone stops ringing because * thinks the phone has been
answered.

Zapata.conf
; Enable call progress
callprogress=yes

Outbound calls to PSTN phone numbers will dial out but there is no
answer detection from the far side. The far side may answer the phone
but * keeps ringing until the timeout expires.

And using this:

exten = 100,hint,SIP/900zap/g0/w5551212

Both the sip phone and zap line both ring at the same time until the
time.
Picking up the sip phone bridges the call and disconnects the zap line
as it should.

Any ideas? We are stuck until after the holidays at this point.
-Kerry



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RE: [Asterisk-Users] Having major issues with TDM2400

[gw]

Hello Kerry, I do it exactly as such, however in steps.  My
understanding of the hint system is just for notification of status, not
for execution of dialing. 

I regularly use this same setup you are looking for, rings in, then
rings 2-5 devices (some zap, some iax) and the first one that answers
gets the call.

Make sure you use the Dial( command I replied with previously. (avoid
hint for testing).

Looking at your emails, it looks like you need to review the dialplan
setup, for example the hint and  do not look right to me.

One example for me: exten =
s,8,Dial(IAX2/ArdsleySomers/314IAX2/ArdsleySomers/331,,)

But it is the same as SIP/220Zap/5, etc.

I cannot say anything specific to amp however.

Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kerry
Garrison
Sent: Sunday, January 01, 2006 7:34 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Having major issues with TDM2400

The goal is to create a user that has a SIP device and a custom ZAP
channel device, have them both ring until one is answered, basically a
ring group.
But I am using AMP's users and device mode rather than the extensions
mode.
I have this working properly on my office system. However, with the
TDM2400 I cannot have both the zap channel and sip channel ringing at
the same time and only handing the call to the end device that answers
the call. I don't understand why this is so difficult for everyone to
grasp. Send a call to both a custom ZAP device and a sip phone and
whoever answers it gets the call.
-Kerry


 

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of C F
 Sent: Sunday, January 01, 2006 4:14 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Having major issues with TDM2400
 
 On 12/31/05, Kerry Garrison [EMAIL PROTECTED] wrote:
  To summarize, I spent 6 hours yesterday on the phone with Digium 
  trying to fix a problem with the TDM2400 ad we still don't have it 
  working right. The lastest version of everything are installed and 
  confirmed by Digium. So here is the issue:
 
  Zapata.conf
  ; Disable call progress
  ; callprogress=yes
 
  Outbound calls to PSTN phone numbers work properly
 
  But using this:
 
  exten = 100,hint,SIP/900zap/g0/w5551212
 
 What are you trying to do here? You trying to hint to a zip channel 
 and dial a number using the hint priority?
 
 
  The extension will ring once, but as soon as the PSTN line
 is picked
  up, the sip phone stops ringing because * thinks the phone
 has been answered.
 
 Which makes sense to me, since as soon as you start dialing you *are* 
 off hook, which in analog means the phone *is* answered. Since all the

 singalling is done in band, it is not difference than picking up the 
 Zap channel for incoming call, at which point you also understand it's

 considered answered.
 
 
  Zapata.conf
  ; Enable call progress
  callprogress=yes
 
  Outbound calls to PSTN phone numbers will dial out but there is no 
  answer detection from the far side. The far side may answer
 the phone
  but * keeps ringing until the timeout expires.
 
 
 So don't use callprogress if it doesn't work for you, in no way do I 
 see this related to the subject line of this post.
 
  And using this:
 
  exten = 100,hint,SIP/900zap/g0/w5551212
 
 
 Again what is this suppose to do?
 
  Both the sip phone and zap line both ring at the same time
 until the time.
  Picking up the sip phone bridges the call and disconnects
 the zap line
  as it should.
 
  Any ideas? We are stuck until after the holidays at this point.
  -Kerry
 
 
 
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RE: [Asterisk-Users] Having major issues with TDM2400

[gw]

Oh just a followup, if you are trying to do an outbound dialout over
analog, what others are saying is correct.  You could consider however
using a voip provider to make the outbound call, then you should have
status.

Greg
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gregory
Wiktor - ADCom Corp.
Sent: Sunday, January 01, 2006 8:05 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Having major issues with TDM2400

Hello Kerry, I do it exactly as such, however in steps.  My
understanding of the hint system is just for notification of status, not
for execution of dialing. 

I regularly use this same setup you are looking for, rings in, then
rings 2-5 devices (some zap, some iax) and the first one that answers
gets the call.

Make sure you use the Dial( command I replied with previously. (avoid
hint for testing).

Looking at your emails, it looks like you need to review the dialplan
setup, for example the hint and  do not look right to me.

One example for me: exten =
s,8,Dial(IAX2/ArdsleySomers/314IAX2/ArdsleySomers/331,,)

But it is the same as SIP/220Zap/5, etc.

I cannot say anything specific to amp however.

Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kerry
Garrison
Sent: Sunday, January 01, 2006 7:34 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Having major issues with TDM2400

The goal is to create a user that has a SIP device and a custom ZAP
channel device, have them both ring until one is answered, basically a
ring group.
But I am using AMP's users and device mode rather than the extensions
mode.
I have this working properly on my office system. However, with the
TDM2400 I cannot have both the zap channel and sip channel ringing at
the same time and only handing the call to the end device that answers
the call. I don't understand why this is so difficult for everyone to
grasp. Send a call to both a custom ZAP device and a sip phone and
whoever answers it gets the call.
-Kerry


 

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of C F
 Sent: Sunday, January 01, 2006 4:14 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Having major issues with TDM2400
 
 On 12/31/05, Kerry Garrison [EMAIL PROTECTED] wrote:
  To summarize, I spent 6 hours yesterday on the phone with Digium 
  trying to fix a problem with the TDM2400 ad we still don't have it 
  working right. The lastest version of everything are installed and 
  confirmed by Digium. So here is the issue:
 
  Zapata.conf
  ; Disable call progress
  ; callprogress=yes
 
  Outbound calls to PSTN phone numbers work properly
 
  But using this:
 
  exten = 100,hint,SIP/900zap/g0/w5551212
 
 What are you trying to do here? You trying to hint to a zip channel 
 and dial a number using the hint priority?
 
 
  The extension will ring once, but as soon as the PSTN line
 is picked
  up, the sip phone stops ringing because * thinks the phone
 has been answered.
 
 Which makes sense to me, since as soon as you start dialing you *are* 
 off hook, which in analog means the phone *is* answered. Since all the

 singalling is done in band, it is not difference than picking up the 
 Zap channel for incoming call, at which point you also understand it's

 considered answered.
 
 
  Zapata.conf
  ; Enable call progress
  callprogress=yes
 
  Outbound calls to PSTN phone numbers will dial out but there is no 
  answer detection from the far side. The far side may answer
 the phone
  but * keeps ringing until the timeout expires.
 
 
 So don't use callprogress if it doesn't work for you, in no way do I 
 see this related to the subject line of this post.
 
  And using this:
 
  exten = 100,hint,SIP/900zap/g0/w5551212
 
 
 Again what is this suppose to do?
 
  Both the sip phone and zap line both ring at the same time
 until the time.
  Picking up the sip phone bridges the call and disconnects
 the zap line
  as it should.
 
  Any ideas? We are stuck until after the holidays at this point.
  -Kerry
 
 
 
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RE: [Asterisk-Users] Having major issues with TDM2400

[gw]

Perhaps a Sipura-3000 could be of use here? Anyone have any ideas about
that?

Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kerry
Garrison
Sent: Sunday, January 01, 2006 10:39 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Having major issues with TDM2400

As much as I like the option of implementing a follow-me type of script,
the second problem is that the client wants to use AMP to manage the
extensions.
Just doesn't seem like I have a solution that fits all of the client's
requirements. The easiest solution seems to be to use a SIP trunk for
the outbound call. 
-Kerry




 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of C F
 Sent: Sunday, January 01, 2006 6:24 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Having major issues with TDM2400
 
 On 1/1/06, Kerry Garrison [EMAIL PROTECTED] wrote:
  Thanks everyone, the reason I posted here was because a
 Digium support
  tech said it should work and he couldn't figure it out. 
 So while I
  appreciate everyone's comments that it wont work, a
 technician from
  Digium said it should, hence I turned to the list for
 clarification. 
  This is not really a good answer for me to go back to my
 client with
  as this is one primary feature he liked which pushed him into an 
  Asterisk solution. For right now,
 
 It will still work using the M option in the dial command, as I wrote 
 before, also look up the follwoing:
 http://www.voip-info.org/wiki-asterisk+cmd+dial
 http://bugs.digium.com/view.php?id=5574
 Using some creativity you can give your client what you promised plus.
 
  their bandwidth is insuffecient for using a SIP provider,
 although a
  T1 line is on order.
 
  -Kerry
 
 
 
 
   -Original Message-
   From: [EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED] On Behalf Of 
   [EMAIL PROTECTED]
   Sent: Sunday, January 01, 2006 5:08 PM
   To: asterisk-users@lists.digium.com
   Subject: RE: [Asterisk-Users] Having major issues with TDM2400
  
   Oh just a followup, if you are trying to do an outbound
 dialout over
   analog, what others are saying is correct.  You could consider 
   however using a voip provider to make the outbound call, then you 
   should have status.
  
   Greg
  
  
   -Original Message-
   From: [EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED] On Behalf Of 
   Gregory Wiktor - ADCom Corp.
   Sent: Sunday, January 01, 2006 8:05 PM
   To: asterisk-users@lists.digium.com
   Subject: RE: [Asterisk-Users] Having major issues with TDM2400
  
   Hello Kerry, I do it exactly as such, however in steps.  My 
   understanding of the hint system is just for notification
 of status,
   not for execution of dialing.
  
   I regularly use this same setup you are looking for,
 rings in, then
   rings 2-5 devices (some zap, some iax) and the first one that 
   answers gets the call.
  
   Make sure you use the Dial( command I replied with previously. 
   (avoid hint for testing).
  
   Looking at your emails, it looks like you need to review the 
   dialplan setup, for example the hint and  do not look
 right to me.
  
   One example for me: exten =
   s,8,Dial(IAX2/ArdsleySomers/314IAX2/ArdsleySomers/331,,)
  
   But it is the same as SIP/220Zap/5, etc.
  
   I cannot say anything specific to amp however.
  
   Greg
  
   -Original Message-
   From: [EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED] On
 Behalf Of Kerry
   Garrison
   Sent: Sunday, January 01, 2006 7:34 PM
   To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
   Subject: RE: [Asterisk-Users] Having major issues with TDM2400
  
   The goal is to create a user that has a SIP device and a
 custom ZAP
   channel device, have them both ring until one is
 answered, basically
   a ring group.
   But I am using AMP's users and device mode rather than the 
   extensions mode.
   I have this working properly on my office system. 
 However, with the
   TDM2400 I cannot have both the zap channel and sip
 channel ringing
   at the same time and only handing the call to the end device that 
   answers the call. I don't understand why this is so difficult for 
   everyone to grasp. Send a call to both a custom ZAP
 device and a sip
   phone and whoever answers it gets the call.
   -Kerry
  
  
  
  
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
 Behalf Of C F
Sent: Sunday, January 01, 2006 4:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Having major issues with TDM2400
   
On 12/31/05, Kerry Garrison [EMAIL PROTECTED] wrote:
 To summarize, I spent 6 hours yesterday on the phone
 with Digium
 trying to fix a problem with the TDM2400 ad we still
   don't have it
 working right. The lastest version of everything are
   installed and
 confirmed by Digium. So 

RE: [Asterisk-Users] Semi-OT: porting numbers away

[gw]

Hello All,
This is in fact some of the best information I have been looking for
regarding porting.

An example is this, I have my primary local numbers on BRI.  I also have
a number of toll free's with a normal ld provider.  What I am doing
right now is pointing my toll-free's to a teliax number.  I have been
very happy with the service Teliax provides, although I wish they were
not using cogent as I get about an 80ms ping from NY  Colorado.  

The thing here is, I am considering porting some numbers over to
teliax(212 numbers, premium 800), not my primary's, but still numbers I
wish to advertise(over 8k/year).  Naturally after advertising I would
always want to be certain I can maintain those numbers.

If for any reason, teliax were to have issues (bankruptcy, etc), what
would happen to these numbers that I may have originally ported?

In the case of toll-free, would it be possible to port those numbers to
another provider relatively easily?  

In the case of conventional numbers, would it be possible to port those
numbers to another provider, or even back to the local provider?

I do not know the legislation, but if a provider simply does not respond
to a port request, does it get automatically approved after like 30
days, or simply nothing happens?

The real question here is, if they were to have issues, how would there
upstream provider react?  Would they allow re-assignment of those
numbers to another downstream provider, or would they just dump them
into a pool and all of my numbers be lost? ( in my case level3 I
believe)

I currently ported my personal numbers to teliax, but that is not an
issue if something were to happen.  On the business side however, it is
an issue. 

Naturally no provider will provide a disclaimer that says 'if we go
under, you are protected', because it makes them sound flaky.  But in
the business voip market, this can be very important.

Of course I trust the local CO will never go under while I am in
business, but with any technology company, since they are not regulated
in the same fashion, does anyone have an idea to what would possibly
happen?  Being down for a few days is not the end of the world, but
losing a premium 800 number for example can be disastrous.

Of course I know the best solution, keep the toll-free on a normal
provider, but there are upsides to using teliax (especially now with
cidname), and it doesn't make much sense to pay for the same service
twice.  The locals can be an important thing though.  I think this is
one of the issues that is effecting the transition of businesses to
VOIP.  I myself have usually suggested against VOIP as a primary in the
business market because of these issues, and aimed more for a hybrid
approach as well, even when connected via T1 or T3.

Thr tough thing is, no company is going to reliably disclose whether
they are profitable or not, and that information can be key.  

On a side note, I have had customers go from Verizon landline 
cablevision  Verizon landline, apparently without too many problems. 
 
Also another off-topic question, does anyone know if it is possible to
port a teliax/level3 number away from teliax/level3?  Such as onto a
cell or to another level3 customer?

Regards,
Greg


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of trixter
aka Bret McDanel
Sent: Friday, December 30, 2005 4:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Semi-OT: porting numbers away

On Fri, 2005-12-30 at 15:41 -0600, Rich Adamson wrote:
   So use call forwarding from the Telco, forward it to a VoIP DID, 
   if you lose the VoIP DID, change the forwarding to another number.
   
  
  I thought my local telco told me that if I were to do that, I would 
  have to pay them LD charges for each call that came in to that
number.
  
  Or am I misunderstanding what you mean by forward here?
 
 Your pstn line will be charge the long distance charge if you 
 forward your local calls to an out of area number.

or have business service that pays per minute.  I worked for an ISP
almost a decade ago that had many residential lines with no services and
call forwarding enabled (total cost less than $10/mo) to use to increase
their dialup numbers.  They forwarded to the main dialup number in the
hunt group.  Largely they were placed at customer sites (in exchange for
discounted service - nondialup customers).  We had 99 forwards enabled,
and becuase they were residential lines local calling meant no
additional cost.  Not a very nice thing to do, but hey after 12 years in
business that isp is still only one county large.  Kinda tells you
something about that ...


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group
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RE: [Asterisk-Users] Hint Priority for Polycom Phones

[gw]

Title: Re: [Asterisk-Users] Hint Priority for Polycom Phones



Hello Doug,
I assume you have subscribecontext set in sip.conf 
right?

Also,I have a 601/sidecar and have the hints 
working fine on the first registration. On my second server registration 
they are not yet coming through.

I am not sure if I can use the hint on the first server 
registration, and have the server point the hint to an iax connection, which is 
what I have started to try doing.

Greg


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas 
GarstangSent: Tuesday, December 06, 2005 11:10 PMTo: 
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: 
[Asterisk-Users] Hint Priority for Polycom Phones

Dang. I must be missing something then. I've modified the contacts 
directory, set bw and bb, can see the buddy on the second appeareance, have 
tried every imaginable combination of the hint command in extensions.conf and 
nada! :( The buddy never updates to show busy/not busy. I thought it was 
interesting too... in the Polycom admin guide it says on page 51 "Notification 
when a change in monitored status will be available in a subsequent release". 
That's for SIP version 1.6.x, dated July 2005. Beats the heck out of me how it 
works when Polycom says it doesn't!

Do you phones send SUBSCRIBE messages to Asterisk on boot? Do you see 
anything if you do a 'sip show subscriptions' for the phone?

Doug

  -Original Message- From: Adam Goryachev 
  [mailto:[EMAIL PROTECTED] Sent: Tue 12/6/2005 
  9:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Cc: Subject: Re: [Asterisk-Users] Hint Priority for 
  Polycom Phones
  On Tue, 2005-12-06 at 21:41 -0600, Jerry Jones wrote: 
  Just in the process of figuring this out myself. i do have it 
  working on an IP601 with a sidecar. Here are my 
  notes. On the polycom Create a contact directory entry 
  for the extension you wish to monitor. Yes the contact must 
  match the exten= statement in your dialplan. Note: It must 
  reside within the same context as the last configured button on 
  your telephone. I have a test phone and had to swap my test 
  extension which is in a test context with my office extension 
  which is in the context with my office phones I wanted to 
  monitor. Had to have the test number register on button one and 
  the office number register on button 2.Nope, this isn't 
  needed... I have an IP600 which registers to asteriskon button 1, another 
  asterisk on button2, a third asterisk on button 3and then has a 
  buddy/hint/monitoring on buttons 4 and 5 which areworking against the 
  first asterisk on button 1. Finally I have a speeddial on button 
  6...Then again, I've not got this working on a polycom IP 300 as 
  yet...Regards,Adam___--Bandwidth 
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RE: [Asterisk-Users] Re: Teliax experiences

[gw]

I also have had good experiences with Teliax. Also 
the CIDName beta function is way cool...

They also offer a pretty plans.

Greg


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk 
Users - DovidSent: Saturday, December 10, 2005 2:54 PMTo: 
[EMAIL PROTECTED]Cc: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Re: Teliax 
experiences

I have been using Teliax for several months now 
with no problems what so ever. However I did have problems with Broadvoice. The 
voice quality isnt allways that great. Sometimes the DTMF wouldt work. It was 
very frustrating when I dialed a company over my Broadvoice line and I tried to 
enter a number and nothing happend. Just my 2 cents.
Regards,Dovid
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[Asterisk-Users] Good Dialing Macros

[gw]

Hello All,
I noticed that AAH seems to have a macro setup that if an extension is
unavailable, asterisk will go auto to menu and such.  It seems to do it
before the dial attempt, so it must be using a macro to obtain the
information.  For example, if a call is forwarded on a phone, it can
skip that phone on an incoming hunt group.

What I am looking for is a script to do the following on HEAD.

Rings in
Check an extension, if busy goto queue, if forwarded take a different
route, and if unavailable bounce to a cell phone.  They key is, I need
to be able to either check the status before the dial, or have it ring
multiple phones  methods, such as an IAX  SIP connection
simultaneously.

Does anyone have a good macro in their dialplan with something like
this?  I don't want to have to load up another aah to reverse engineer
it...

Searched the wiki but turned up nothing...

Regards,
Greg
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RE: [Asterisk-Users] echo canceling algorithm

[gw]

Hello Joe,
I asked the same question. It is probably a 
combination of things, hardware issues like PCI bus latency, plus an issue of 
interest. I suspect the big companies have managed to cover this, but also 
because they charge a lot for their equipment.

Being open source, yes you could write your own, and their 
probably are bounties out there right now (of which I would add to). I am 
going with a hardware tellabs can via a channel bank. A real 
solution? I asked this a while ago on the list, and my opinion is it 
probably will not happen within the next 3-6 months. After that I would 
not know either, but the popularity of asterisk has grown so much that possibly 
we would get lucky with some good algo's that cover more diverse 
situations.

I find that digium has a great product, and the echo is a 
big issue. Many people can usually tweak it out, some cannot (especially 
those with issues on lines or long loops). Even if you are on a long loop 
that does not mean the $1000 card will be perfect. I spent more than that 
on a bank and t1 card, but there is also a lot of flexibility. 


If you are running 12 ports I would certainly consider a 
channel bank or one of those digium cards (which is actually less than your 
three tdm400's I beleive), but keep in mind that using a hardware EC like a 
tellabs requires either a shelf, or some good project 
skills.

The aggressive works well, though I cannot say it is 100% 
perfect.

Maybe a pci-express digium card would help :) Just an 
idea.

Greg


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Joe 
PukepailSent: Friday, December 02, 2005 6:28 PMTo: 
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] echo canceling algorithm

I have been wondering about echo canceling, it seems to be one of the major 
problems people have with asterisk. I've gotten it to acceptable levels 
(using mark2 aggressive), but everything I've read indicates that the echo 
canceling software isn't very effective. 

My question would be, what do we need to get an effective echo canceler (in 
asterisk software)? Is it patent issues? No experience (I know I 
don't know anything about how to write a echo can algorithm) or just getting the 
right people interested in writing one ($)? 

With digium offering hardware echo can, I can only conclude that echo can 
can't be done effectively in software? If it is a matter of money perhaps 
a bunch of users can offer bounties for someone (or some company) to write an 
good echo canceler? 

With the amount of money that a hardware echo canceling card costs (+$1000 
per T1/E1) if half of this were spent on a fund for software echo cancel it 
would seem we could do it (if it is even possible using todays technology??). 


I don't mean this as critical of the developers who have done so much, just 
an honest question what we (as users) can do to help improve the product. 

On 12/2/05, Patrick 
Fortin [EMAIL PROTECTED] 
wrote: 
HiJust 
  wandering what solution worked to eliminate echo on your setup.I am 
  trying every solutions I can find on the wiki and none is working 
  perfectly.We have asterisk 1.2.03 x digium TDM400P30 
  Snom320 + 5 Snom360For now the best setup I have is using Mark2 Echo 
  cancel.ThanksPatrick___ 
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[Asterisk-Users] Voicepulse not accepting new customers. (FCC E911)

[gw]

Hello All,
It seems that voicepulse is not taking any new orders on the standard
service plans (though vp connect seems unaffected) due to the fcc
rulings.

We'll see what happens, anyone having similar problems with other
services as of today?

Greg
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RE: [Asterisk-Users] CallerID not passing through to Polycom 500

[gw]

 
Check your logs, make sure you are waiting long enough before sending
the call to the polycom.

Uf asterisk sees the CID, it should send it and it should show up on the
polycom.

Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary
MacKay
Sent: Thursday, November 24, 2005 11:19 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] CallerID not passing through to Polycom 500

I have a basic system working, except for callerid. The Polycom 500 just
shows call from Business Line on the screen. Business Line is the
name of the context that line is in. How do I get it to show the
callerID on the screen instead? Yes, I have CallerID on that line and it
works on a standard analog phone.
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RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk install

[gw]

I do believe there is a system reset is there not? Thought I saw it in
the manual.

Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Thursday, November 24, 2005 8:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk
install

Did you get it?  I would like to take a whack at it if not.



 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, November 23, 2005 10:30 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk

 install
 
 Does anyone know of a brute force that will work on a serial interface

 like hyperterminal?
 
 --Jim
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
 Sent: Wednesday, November 23, 2005 8:22 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Help need to reset Adit 600 for Asterisk

 install
 
 Is the password limited to four digits like the Adtran 600 (I think)?
 
 Start plugging in numbers.  Only 10,000 possible combinations.
 
 
 
  -Original Message-
  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
  Sent: Wednesday, November 23, 2005 9:59 AM
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Subject: RE: [Asterisk-Users] Help need to reset Adit 600 for
Asterisk
  install
 
  Thanks Jerry,
 
  I have called Carrier Access and they can reset the password but for
a
  considerable fee.   We have serial access but after it boots it
  immediately
  asks for a username and password.  We have the username but the
 password
  is
  not what it is suppose to be.   There's a reset switch on the
 faceplate
  but
  I think the LOCAL SET is OFF and that is why it doesn't respond.
 Their
  manual says the Reset switch is not under the control of LOCAL SET,
 yet it
  doesn't seem to work.  Well, we might not know the proper boot
 sequence.
  It
  contains flash memory and there is a timing that important to that
 reset
  procedure.  Anyone's help is much appreciated.
 
  --Jim
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of Jerry
 Jones
  Sent: Wednesday, November 23, 2005 7:40 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [Asterisk-Users] Help need to reset Adit 600 for
Asterisk
  install
 
  Not sure but are you connecting via serial or ehternet? Seems to be
 the
  serial had a way to do this easily on bootup. Otherwise I would be 
  interested for future reference. Carrier Access does have a good
 support
  team, just need to know your serial number.
 
  On Nov 23, 2005, at 12:13 AM, [EMAIL PROTECTED] [EMAIL PROTECTED]
 wrote:
 
   Looking for a way to hard reset a ADIT 600 just purchased used.
   But it
   seems to have a master password already set.  We've tried the
front
   reset but maybe we don't have the right sequence of boot order.
Any
   help would be much appreciated?  - Jim
  
  
  
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RE: [Asterisk-Users] Eicon Diva Server query

[gw]

I would go with chan-capi-cm, as well as loading up the eicon drivers
first for the base drivers and utility set.

I have a few installations as such that are working flawlessly, and
Armin has done great work on the driver.

Regards,
Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Avi Miller
Sent: Wednesday, November 23, 2005 3:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Eicon Diva Server query

David Waugh wrote:
 Yes, you can use the Eicon Diva Range with 2.6 Kernels

Another question, considering the card should arrive tomorrow and I'd
like to try my hand at setting it up this weekend: Do I need to BRIstuff
Asterisk to get the Eicon Diva V-4BRI to work, or should I just need
chan_capi-cm?

Thanks,
Avi

--
National Manager - Special Projects

 Sydney / Melbourne / Canberra / Hobart / London /
   Walter Turnbull Bldg   T: +61 (0) 2 6233 0607
   44 Sydney Ave, F: +61 (0) 2 6233 0696
   Forrest,   W: http://www.squiz.net/
   ACT 2603

. Open Source  - Own it  -  Squiz.net ./
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RE: [Asterisk-Users] VOIPJET - are they down

[gw]

There could be something funny going on today, in the area of call
completion.

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph
Sent: Friday, November 18, 2005 4:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VOIPJET - are they down

On Fri, 2005-11-18 at 11:40 -0800, Luki wrote:
  Can anybody confirm if there is a problem with their server.
 The east cost server I use (64.34.45.100) works fine.
 
 --Luki

That is strange, I can not make a out either through Teliax or VOIPJET.
Calls via FWD are working fine.

--
#Joseph
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RE: [Asterisk-Users] CVS v1-2-0 make problems?

[gw]

Do a search on this list, there is a fix for this. Rare but can happen.

Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kristian
Kielhofner
Sent: Thursday, November 17, 2005 3:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] CVS v1-2-0 make problems?

Asterisk-users,

Has anyone else had problems with the v1-2-0 CVS rev?  Here's
the deal:

LATE last night I checkout out 1.2.0 with CVS:

rm -rf asterisk zaptel libpri
cvs co -r v1-2-0 zaptel
cvs co -r v1-2-0 libpri
cvs co -r v1-2-0 asterisk

zaptel and libpri build fine.  Asterisk, however, seems to get
stuck in a infinite loop while (guessing) determining version.  The loop
occurs when using cmp to check version.h and version.h.tmp.  It goes on
forever, forever, and forever.

However, using the 1.2.0 tarballs work perfectly, for libpri,
zaptel, and asterisk.

Yes, this is for AstLinux and it is using my cross-build
environment. 
(Which has worked very well for tracking CVS HEAD at build.astlinux.org,
and as mention before can build using the 1.2.0 tarballs).

I'd have more time to dig deep into it, but I am just trying to
get a
1.2 build of AstLinux done.  I somewhat foolishly promised one by
tomorrow :).

Anyone else experiencing this?  Are my CVS commands wrong?
What's up?

Thanks in advance, and a HUGE thank you to everyone at Digium for
getting 1.2 out!

--
Kristian Kielhofner


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RE: [Asterisk-Users] RE: open asterisk?

[gw]

And development costs?  To develop a card is not cheap, unless you are
in China :) 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, November 16, 2005 3:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] RE: open asterisk?

10 Million is ca 10,000 boards/licenses if you assume 1000 USD in
average. I know the Digium boards are some of the cheapest around, but
the actual production cost should be below 150 USD for a 4xE1/T1 of this
type, meaning that they still should have very decent margins.

jan

[EMAIL PROTECTED] wrote:

I wouldn't believe 10mil/yr.   Maybe if they had other non-asterisk
products, but that just does not seem reasonable if you look at 
asterisk at this stage.  Besides, are they a public company? Are they 
required to report to the SEC?  That makes a big difference.  I mean 
realistically, a 900 dollar software product or less than 10k hardware 
product needs a lot of sales to hit around 10mil.  And the tech support

would have enourmous overhead.

But if digium is also doing custom ivr's and such, you can easily do a 
few million there, however you have to have well over 100 employees.

Oh unrelated, I have talked to people who tend to hype up their
company.
I feel it is bad business, but have heard people saying 'yes I will hit
5 million this year, and have no loss'.  Later, I have seen them go 
under.

Greg


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of calvis
Sent: Tuesday, November 15, 2005 10:27 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] RE: open asterisk?

Maybe this will open your eyes :)

This article says Digium has sales of 10 million per year.


http://news.com.com/Is+the+telephone+industry+ready+for+open+source/200
8
-108
2_3-5737703.html



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of 
[EMAIL PROTECTED]
Sent: Tuesday, November 15, 2005 7:04 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] RE: open asterisk?

Millions of dollars in hardware sales?  I admit I don't know what there

sales level is, but I doubt that it's 1 mil at this stage in the game.
And, if it were, that has nothing to do with profit.  You can make 1 
mil in sales in a year but still walk away with a net loss.

Without the community where would digium be? Don't know, probably doing

the same but in a proprietary format, giving people no choice but to 
shell out the big bucks.

I for one am happy they started up the project, and if they did in turn

make Asterisk proprietary, I would probably shell out the cash since it

offers what no other pbx can offer.

Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aidan Van

Dyk
Sent: Tuesday, November 15, 2005 8:51 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] RE: open asterisk?

[EMAIL PROTECTED] wrote:

  

As for the people who would suspect digium is strong-arming anyone, 
hell, if it weren't for them you wouldn't have asterisk would you?
And therefore probably no openpbx either, and we all would be spending



  

thousands to do what asterisk can do for free.



And if it weren't for the community of indentured slaves and testers, 
where would Digium be, with no users, contributors, or bug-reporters?
Never mind the no millions of dollars of hardware sales.  It *should* 
be a symbiotic relationship.



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RE: [Asterisk-Users] 2 SIP phones on Y data connector on 1 ethernet

[gw]

 Unless, if this is what you mean, can you use one cat5 wire to run two
home runs?  There are these adaptors that allow two signals by
merging/splitting 12,36 as the first feed, and 34,78 as the second.
This would work, but don't plan on going gigabit.  Personally I prefer a
hub or switch, but in some cases you cannot do this.  Also POE is out of
the question on such a setup.

Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
Horan  Company, LLC
Sent: Tuesday, November 15, 2005 7:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 2 SIP phones on Y data connector on 1
ethernet

Actually, that's not specifically true - a hub of theoretically any size
could be made just by carefully and accurately connecting together the
tx and rx pairs of a bunch of rj45 cables together.  I don't think the
wire gauge would allow for more than a couple nodes though.  I've got a
bud who says he did this in school - all tx pairs must connect to all rx
pairs.  Up to the cards for collision detection :)

Moj

Humberto Aicardi wrote:
 A_ Navone,
 
 You cannot use a Y connector on a data (ethernet) connection, you must

 use a switch or and older hub to accomplish this.
 
 Regards,
 Humberto
 
2 SIP phones on Y data connector on 1 ethernet - will that cause 
problems ?
thx in advance

_
Don't just search. Find. Check out the new MSN Search! 
http://search.msn.click-url.com/go/onm00200636ave/direct/01/

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--
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Office Manger, Horan  Company, LLC
(907) 747- x112
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RE: [Asterisk-Users] RE: open asterisk?

[gw]

Millions of dollars in hardware sales?  I admit I don't know what there
sales level is, but I doubt that it's 1 mil at this stage in the game.
And, if it were, that has nothing to do with profit.  You can make 1 mil
in sales in a year but still walk away with a net loss.

Without the community where would digium be? Don't know, probably doing
the same but in a proprietary format, giving people no choice but to
shell out the big bucks.

I for one am happy they started up the project, and if they did in turn
make Asterisk proprietary, I would probably shell out the cash since it
offers what no other pbx can offer.

Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aidan Van
Dyk
Sent: Tuesday, November 15, 2005 8:51 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] RE: open asterisk?

[EMAIL PROTECTED] wrote:

 As for the people who would suspect digium is strong-arming anyone, 
 hell, if it weren't for them you wouldn't have asterisk would you?  
 And therefore probably no openpbx either, and we all would be spending

 thousands to do what asterisk can do for free.

And if it weren't for the community of indentured slaves and testers,
where would Digium be, with no users, contributors, or bug-reporters?
Never mind the no millions of dollars of hardware sales.  It *should* be
a symbiotic relationship.



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RE: [Asterisk-Users] Cisco 7960 Multiple Line Appearance

[gw]

Why not have set lines 3-6 as separate sip registrations, and have
asterisk ring multiple phones? 

Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sergio
Chersovani
Sent: Tuesday, November 15, 2005 9:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Cisco 7960 Multiple Line Appearance

Matt Hoskins ha scritto:

 I'm currently using asterisk 1.0.7 with cisco 7960 SIP 7.5 phones.  
 I'd like to use this phone for a receptionist so that she can take 
 calls for  4 other people.  Is this possible?

The SIP firmware does not support it.
You have to use SCCP to do that

 Is there any way to do this with SIP and the 7960?  I've seen the 7914

 but then I'd have to use SCCP and I'm not sure if it is stable enough 
 for production use.

Well give it a chance :-)

http://chan-sccp.berlios.de

Sergio
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RE: [Asterisk-Users] Problem with call drops

[gw]

What kind of switches?  I would suggest go gigabit between your servers,
their switch, and the call center floor.  And, use managed switches if
you are over 20 total stations.  Could be someone's got a p2p and it is
killing you.  Maybe unlikely, but quite possible.  Use some good 3com
switches too, and totally avoid hub and daisy chaining.  Or get a
managed switch with expansion switches if you are running a large number
of ports.  Either way, a single cable drop is a very bad idea.  You get
a snagged cable or something and you can be out of business because of a
stupid cable.

Regards,
Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Waldo
Rubinstein
Sent: Tuesday, November 15, 2005 1:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Problem with call drops

Hello guys,

I've been having a a recurring problem with people complaining about
calls being dropped.

I have 3 asterisk servers:

Gateway: running Asterisk 1.2rc2 with TE410P connected to 4 T1s of the
PSTN Server 1: running Asterisk 1.2rc2 with ztdummy using Gateway to
access the PSTN (in/out) Server 2: running Asterisk 1.0.9 with ztdummy
using Gateway to access the PSTN (in/out)

I have agents connected to both Server 1 and Server 2.

The problem is that every once in a while, agents complain that their
calls dropped. Sometimes it's some agents that complaint, some other
times, ALL agents complaint at the same time.

When these complaints occur, the only thing I so is:

1) Issue a show channels on all three servers to see if there are any
ongoing calls. I assume that if there are active calls, that not all
calls dropped
2) Issue dmesg in Gateway to see if any of the T1s presented an alarm
that could have dropped the calls

In either case, I don't see any errors in dmesg indicating there were
problems with the T1s and I always see active calls in all the servers.

What can I do to further troubleshoot this? If I look at the CDR, how
can I tell that a call was abnormally terminated? Are there any tools
out there that would allow me to check the health status of the working
system and not just the availability of the machines? I was looking at
nagios but I don't know if it will do what I need to do.

The other thing I could think of is that there may be a network problem
in the building. The way everything is connected is all agents are
connected to a series of unmanaged switches in the call center floor,
which are daisy chained together and then one is connected to the main
switch with a single cable drop. The main switch is where the Asterisk
servers are connected. I figured that if one of the switches in the call
center area fails, it could drop all calls or if there is a problem with
the cable drop from the main switch, that could also affect all agents.

Any advise will be greatly appreciated.

Thanks,
Waldo
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RE: [Asterisk-Users] Has anyone bought anything from Asteriskmall? yourexpirence?

[gw]

Their tdm24 prices are lower, but many others like channel banks and
te405 are higher than voipsupply.

Just shop around...  Sorry though, I have not dealt with them before.
Voipsupply.com has always been good with me...

Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John
Reynolds
Sent: Tuesday, November 15, 2005 1:24 PM
To: Asterisk Users Mailing List
Subject: [Asterisk-Users] Has anyone bought anything from Asteriskmall?
yourexpirence?

I placed an order with asteriskmall.com ... but am wondering if I have
made a grave mistake.

Wondering if anyone has had any expirence with these people, of if it is
just some scam site.

You input is appreciated.

John
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RE: [Asterisk-Users] RE: open asterisk?

[gw]

I wouldn't believe 10mil/yr.   Maybe if they had other non-asterisk
products, but that just does not seem reasonable if you look at asterisk
at this stage.  Besides, are they a public company? Are they required to
report to the SEC?  That makes a big difference.  I mean realistically,
a 900 dollar software product or less than 10k hardware product needs a
lot of sales to hit around 10mil.  And the tech support would have
enourmous overhead.  

But if digium is also doing custom ivr's and such, you can easily do a
few million there, however you have to have well over 100 employees.

Oh unrelated, I have talked to people who tend to hype up their company.
I feel it is bad business, but have heard people saying 'yes I will hit
5 million this year, and have no loss'.  Later, I have seen them go
under.

Greg


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of calvis
Sent: Tuesday, November 15, 2005 10:27 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] RE: open asterisk?

Maybe this will open your eyes :)

This article says Digium has sales of 10 million per year.


http://news.com.com/Is+the+telephone+industry+ready+for+open+source/2008
-108
2_3-5737703.html



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, November 15, 2005 7:04 PM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] RE: open asterisk?

Millions of dollars in hardware sales?  I admit I don't know what there
sales level is, but I doubt that it's 1 mil at this stage in the game.
And, if it were, that has nothing to do with profit.  You can make 1 mil
in sales in a year but still walk away with a net loss.

Without the community where would digium be? Don't know, probably doing
the same but in a proprietary format, giving people no choice but to
shell out the big bucks.

I for one am happy they started up the project, and if they did in turn
make Asterisk proprietary, I would probably shell out the cash since it
offers what no other pbx can offer.

Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aidan Van
Dyk
Sent: Tuesday, November 15, 2005 8:51 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] RE: open asterisk?

[EMAIL PROTECTED] wrote:

 As for the people who would suspect digium is strong-arming anyone, 
 hell, if it weren't for them you wouldn't have asterisk would you?
 And therefore probably no openpbx either, and we all would be spending

 thousands to do what asterisk can do for free.

And if it weren't for the community of indentured slaves and testers,
where would Digium be, with no users, contributors, or bug-reporters?
Never mind the no millions of dollars of hardware sales.  It *should* be
a symbiotic relationship.



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RE: [Asterisk-Users] TDM Echo issue

[gw]

Hello Sacha,
While it is not the best solution as far as quality is 
concerned, I would suggest you at least try the aggressive canceller in 
zconfig.h. I put it into use temporarily while I get an external echo can 
setup. It takesa bit of getting used to (no simultaneous 
speech/duplex), however it's really not bad if you are on a long loop from your 
CO (a cause of many troubles).

Txgain -4.5 seems low to me, but it all depends on your 
lines.

Also, be careful of the locations of the gain settings, 
they need to be within the channel definition. If you are mixing gains on 
different lines (like to an ATA or PBX where they are 0), you need to be careful 
about the config file.

Greg


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Sascha 
FerleySent: Monday, November 14, 2005 12:09 AMTo: 
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: 
[Asterisk-Users] TDM Echo issue


Hi, 

I am running into a issue with a 
TDM04B card. When dialing out I get an noticeable (extreme to some people) echo, 
in that I can hear myself. The person on the other line doesnt hear any echo 
and the call sounds perfect to them. 

I checked and tested a few things as 
per suggestions on voip-info.org with RX/TX gain and using ztmonitor. I adjusted 
the 
rxgain=10.5 txgain=-4.5 and it 
doesnt seem to do to much to eliminate me hearing myself on the phone. I 
cant go any lower on txgain then -5.5 before the call doesnt go through any 
more. If I change the txgain to above 0 the echo gets even worse. 


I am using Cisco 7960 phones and 
calling IP to IP is perfect; the echo occurs only when going out the zap 
channels to the PSTN. Below is the relevant zapata.conf file. I also 
checked the /proc/interrupts file and the interrupts seem normal (see below). 


If anyone has any other suggestion, 
please let me know,

Thanks

Sascha


### /proc/interrupts 

 
CPU0  
CPU1
 0:  
62224232  62229547 
 
IO-APIC-edge 
timer
 1:  
0 
 
3 
 
IO-APIC-edge  
keyboard
 2:  0 
 
0 
 XT-PIC 
 
cascade
 4:  
0 
 
5 
 
IO-APIC-edge  
serial
 8:  
0 
 
1 
 
IO-APIC-edge  
rtc
14:  
0 
 
2 
 
IO-APIC-edge  
ide0
18:  
556800  
715820  
IO-APIC-level  
libata
20:  562694111 
 681819012  IO-APIC-level 
 
wctdm
53:  
24008833  8 
 
 
IO-APIC-level  
eth0
NMI:  
0 
 
0
LOC:  123628432  
123628430
ERR:  0
MIS: 0


# /etc/asterisk/zapata.conf 
#
;
; Zapata telephony 
interface
;
; Configuration file

[trunkgroups]

[channels]

language=en
context=from-pstn
signalling=fxs_ks
rxwink=300 
; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO 
lines
;
;usedistinctiveringdetection=yes

usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
;echotraining=800
echotraining=yes
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no

;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no

;Include genzaptelconf 
configs
#include zapata-auto.conf

;Include AMP configs
#include 
zapata_additional.conf

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RE: [Asterisk-Users] ISDN card required

[gw]

Yes, but if you are in the states you need the eicon for support of 5ess
and dms-100.

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Elkins
Sent: Monday, November 14, 2005 6:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] ISDN card required

So far - the 4-port ISDN HFC chipset cards from Junghanns.net works well
and is half the price of a 4-port Eicon card.


On Mon, 2005-11-14 at 10:07 +, David Waugh wrote:
 Hi Lee,
  
 I use a Diva Server card here with Asterisk using Chan_capi.
 The basic BRI card has one BRI port. They also have a model with 4 
 port BRI model. You can mix and match Diva Server card too, so as your

 needs expand you can add more cards to your server.
  
 Further information can be found on the Eicon website:
  
 http://www.eicon.com/worldwide/solutions/Diva_Server_and_Asterisk
  
 and
 http://www.eicon.com/worldwide/products/MediaGateways/all-in-one.htm
  
 Thanks
 David
  
  
  
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of
 Lee Archer
 Sent: 14 November 2005 09:32
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] ISDN card required
 
 
 
 Can anyone point me in the direction of a quality, works with
 Asterisk, BRI card.  I need minimum 2 port/4 channel. 
 
 Regards
 
 Lee
 
 ###
 
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 Microsoft Exchange.
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-- 
  .  . ___. .__  Posix Systems - Sth Africa.  e.164 VOIP ready
 /| /|   / /__   [EMAIL PROTECTED]  -  Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS  Tel: +27 12 807 0590  Cell: +27 82 601 0496

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RE: [Asterisk-Users] open asterisk?

[gw]

Haven't we all lost sight of something here?  Allison is a person, not
an object.  If she wants to do work for one person rather than the
other, it is up to her.  Code is a different story sure, but this is a
human being you are talking about.

This is a free enterprise, if you want to have a project of your own,
get your own people.  Allison's voice may be popular within the asterisk
community, but personally I would rather have a voice like James Earl
Jones' as Verizon has, or Majel Barrett-Rodenberry.  Of course, they are
top names and would not do such things unless serious money were
involved.

A person does what is in their best interest, and there is nothing wrong
with that.

I am not defending one person or another, but people need to be
realistic here.  I know not what the contractual arrangements are, but
agree that open source has nothing to do with a person's voice,
personality, or any other aspect of a person.  Open source is about
sharing knowledge.

As for the people who would suspect digium is strong-arming anyone,
hell, if it weren't for them you wouldn't have asterisk would you?  And
therefore probably no openpbx either, and we all would be spending
thousands to do what asterisk can do for free.

Regards,
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jon
Pounder
Sent: Monday, November 14, 2005 8:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] open asterisk?




Guys, this arguing and choosing sides is starting to get a little like
American Hot Rod. If I wanted to listen to that nonsense I'd just tune
it in.

Code contribution is one thing, but by now I would think this bridge
would have been crossed with voice files. Sure Allison is probably
cleaning up doing a little bit here and there for anyone who needs
something small added to a stock setup, BUT, is this really practical
for a business ?
There should be simple application that can be used to rerecord and
substitute all current voice files in a system.

There are plenty of ways a catalog could be kept of files and the exact
contents of them, including right in the files themselves. mpeg, and wav
files both provide a means for storing other information right in with
the audio, so why not use this ? Then a simple manager application could
be used to find all the files and generate a script to be used to
recreate them, a few auditioning and approval menu functions and then
the ability to replace the existing set with the new set all at once
would make a pretty complete and stand alone application.

No matter which telephony project is involved the voice content should
not hinge so directly on one person and whatever their allegences or
other problems might be.


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RE: [Asterisk-Users] Quantumvoice vs Broadvoice - Multiline

[gw]

Have you looked into teliax?  4 simultaneous calls on a bus plan is
pretty good for less than $50/mo. And I cannot complain about the
quality or the support.

Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Saul Diaz
Sent: Friday, November 11, 2005 4:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Quantumvoice vs Broadvoice - Multiline

Julio Arruda wrote:


 I was testing Broadvoice few weeks before Hurricane Wilma here in FL.

 Since then, I had been since the landline (Bellsouth), and I had to 
 'remote callfwd' the BS # to my broadvoice #.

 So, from my impression, is ok for my needs (I got a weird no ringback 
 problem that I kind of solved with a Background trick), and no 
 surprises yet regarding the bill (my mother in law call Brazil a lot 
 from my house, no, she is not aware of the 'unlimited' plan. So I may 
 be in for a surprise in a couple of months).
 I've no tried several calls at the same time, you may want to ask
them..
 PS: I'm running Asterisk 1.0.9

 Dane Reugger wrote:

 We are considering Quantumvoice as a provider -

 They are telling us they will give us 1 line number but we can have 5

 concurrent incoming and outgoing line numbers. Charge is about $45 + 
 extras - this seems considerable less expensive than the competition 
 which seem to focus on.

 My second choice is BroadVoice $29.99 + $9.99 per additional line (in

 state only?) - more expensive, less features, and they don't seem 
 loved by many ?

 Is anyone else using Quantum Voice?
 It was mentioned earlier that it requires an ATA connection and 
 Asterisk support/compatibility is sketchy at best - I've contacted BV

 and they responded saying they need 24hrs to look into it?

 Seems like a popular topic but I'm looking for 2-3 lines - I only 
 need one number but need to be able to make or receive several calls 
 at a time?

 Any advice or recommendations appreciated - I want to port  my number

 but I'm running out of time and must make a decision very soon.


 Thanks,
 Dane Reugger
 Crescent City Technologies
 New Orleans, LA 70112
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Broadvoice only allows only the normal 3 way calling so is 2 channels
for #

about BV i got a lot of water under the bridge every works ok supper
ok for times. then BV brokes without you make a single change in your
asterisk server and stop working.. if u call support you are the guy
with the problem.. yes BV support sucks, and it took me 9 phone calls,
12 emails, 3 chargeback and 2 call to my bank to remove myself from
their billing all them well documented...

so my advice nothing can be worts than BV.

regards
Saul
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RE: [Asterisk-Users] RE: (BAD!!!) Sound quality of the NEW GRANDSTREAM BT 101 and 102 MODEL

[gw]

I just got a new SPA-941. Awesome phone for the 
price, worth looking into...

Greg


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of André 
Rodrigues ( Cheyenne)Sent: Thursday, November 10, 2005 4:17 
PMTo: 'Asterisk Users Mailing List - Non-Commercial 
Discussion'Subject: [Asterisk-Users] RE: (BAD!!!) Sound quality of 
the NEW GRANDSTREAM BT 101 and 102 MODEL


Does 
anyone have tried these new grandstream BT 
models?

Don’t 
you have the same problem when you test a new one compared with an old 
one?

Please help me. I need to by more than 50 phones and the test phones are 
from the new model and they are not working well.

I’m using 
firmware 1.0.6.7 and I’m making final 
tests whit old BT models and the new ones, with the same firmware and version 
1.0.518 to, and my Client does not want to approve the solution with this 
problem on the new model. 

I don’t 
have any problem with the Gxp model.

BT sound 
quality has no “cuts” or noise. But the sound is much more “lower” and not clear 
and crystalline. I’m using PCMA and I have tried several codecs. If I only turn 
on the speakerphone the line sound on an “old” BT is much higher than on a new 
one…

Regards
André


De: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] Em nome de André 
Rodrigues ( Cheyenne)Enviada: quinta-feira, 10 de Novembro de 2005 
17:34Para: asterisk-users@lists.digium.comAssunto: 
[Asterisk-Users] (Some problems sending this menssage) Soundquality of the new 
BT 101 and 102 models Importância: Alta






Hi.


I’m 
having sound quality problems using the new BT 101 and 102 models (the ones with 
solid colour bottoms like the gxp model). I’m using firmware 
1.0.6.7.

Does 
anyone as the same problem with these new 
models?

Sound 
quality has no “cuts” or noise. But the sound is much more “lower” and not clear 
and crystalline. I’m using PCMA.

Regards.

André 
M. S. Rodrigues
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RE: [Asterisk-Users] sill looking for a provider

[gw]

Does it say I use them?  I only said that voipjet comes through at 19ms,
so I disagree about the TOS. (didn't know about it anyway :)

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, November 07, 2005 5:42 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] sill looking for a provider

OOPPS!  Looks like someone just broke voipjet's tos

gw at adcomcorp.com gw at adcomcorp.com wrote on Sat Nov 5 11:36:46 CST
2005 




 I tend to agree with you, my experience with Teliax has been decent,
and getting better.  If only I could get to them at under 20ms though,
right now my latency is about 75ms whereas voipjet comes through at
19ms.

Greg

--

https://www.voipjet.com/tos.php
NON-DISCLOSURE: ALL CUSTOMERS USING VOIPJET'S SERVICE ARE SPECIFICALLY
PROHIBITED FROM DISCLOSING TO OTHERS THAT THEY USE VOIPJET'S SERVICE,
THIS INCLUDES BUT IS NOT LIMITED TO, END USERS. CUSTOMERS MAY NOT
DISCLOSE USE OF OR PAYMENTS TO VOIPJET ON PERSONAL, CORPORATE, LEGAL,
ACCOUNTING AND OTHER DOCUMENTS AND COMMUNICATIONS UNLESS SPECIFICALLY
REQUIRED TO DO SO BY LAW


Has anyone else read these TOS'es???  Some are pretty funny.


Thomas Herlihy
Scaletta Moloney Armoring
Chicago, IL USA
708.924.0099
Skype VoIP @ HerlsOne
Free World Dialup 647717
[EMAIL PROTECTED]
www.scaletta.com
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[Asterisk-Users] CallerID via chan_capi-cm-0.6 possible?

[gw]

Hello All,
I have a bri and iwsh to get CID w/name, however, even though Verizon
has told me that CID/Name is on the circuit, I still only get ANI.  No
cid or cid/name.

Anyone know if it is possible to get cid over bri?

I am not sure if the issue could be in the eicon firmware or something
else, since the eicon logs don't mention the CID or CIDName...

Btw it is on a DMS100 switch.

Regards,
Greg
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RE: [Asterisk-Users] sill looking for a provider

[gw]

 I tend to agree with you, my experience with Teliax has been decent,
and getting better.  If only I could get to them at under 20ms though,
right now my latency is about 75ms whereas voipjet comes through at
19ms.

Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cullin J.
Wible
Sent: Saturday, November 05, 2005 8:51 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] sill looking for a provider

We have been using Teliax (www.teliax.com) for a while now and have 3
accounts with them (one for each of our asterisk servers). They've had
their ups and downs but have been working to improve their support and
now we are now able to speak with someone during their business hours
(8-5PM MST). 

There was recently an issue with a high-latency link between our
backbone and their backbone providers. We called them about it to find
out that they had already escalated the issue with the backbone carrier
and the issue was resolved. The long and short of it is that when we
have had issues (which are few and far between) they have been quickly
resolved.

There is no question in my mind that a growing industry and growing
companies are going to face customer service issues. It's bad for us
now, but I think it's a good sign for the open source VoIP community. I
think patience is key.

Originally we looked at a number of carriers, based on the following
requirements:

1) Be located in the US.
2) Have a customer support phone number and answer the phone.
3) Accept major credit cards and automatically bill my account (no need
to recharge via pay-pal).
4) Allow for business/corporate usage.
5) Support IAX and g726/g729 codec's
6) Support Set Caller ID
7) Support multiple-inbound DID's

We didn't care about call forwarding, voicemail, 3-way calling or any of
the other features that must residential carriers tout as features. I
wish I had done my research a but more formally, but the answer after
about a week of research and test accounts was to use Teliax.

Hope that helps.

Cullin J. Wible
President  CEO
Algorim Technologies, LLC
212-535-3238 x102
[EMAIL PROTECTED]
 
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Piotr A.
Sygula
Sent: Friday, November 04, 2005 6:44 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] sill looking for a provider

That concept is not bad; except when the CEO from the same company as
the tech that calls all the time happens to call you from what appears
to be the same caller id, and the CEO ends up hearing rap or hard
rock...


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Sent: Friday, November 04, 2005 5:32 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] sill looking for a provider

Jason Brashear wrote:

 Is there a provider that has good support and answers the phone? (=

 I need to get lines for my Asterisk server and want to move from 
 broadvoice.com.

 So far I haven't been able to get anyone on the phone.

 Too funny...

I was able to get them on the phone today but it means waiting on hold a
very very long time.

Maybe I should look for a provider that uses good quality comedy instead
of music on hold?

Even better we could add a feature to asterisk where you set your
preference. Press 1 for rock, 2 for rap, etc. and the system uses your
caller ID to remember that for subsequent calls.

The latest acronym is the industry is HOIP. That stands for hold over
IP. Rumors are that it will be patented in the US soon.

You've been a great audience. Thank you very much.


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[Asterisk-Users] 10/28 head 10/29 head capi issue

[gw]

Hello all,

On HEAD 10/28/2005 my chan_capi-cm-0.6 is working fine.  If I go to
10/29/2005 or newer, something freaks out and I get the following
behavior:

*CLI   == ISDN1: Incoming call '19142775896' - '2781980'
-- Executing Set(CAPI/ISDN1/2781980-0, IncomingLine=2781980) in
new stack
-- Executing NoOp(CAPI/ISDN1/2781980-0, 19142775896) in new
stack
-- Executing Set(CAPI/ISDN1/2781980-0, CALLERID(name)=PUMS IN
TollFree) in new stack
-- Executing Set(CAPI/ISDN1/2781980-0, CALLERID(num)=9142775896)
in new stack
-- Executing Set(CAPI/ISDN1/2781980-0,
CALLFILENAME=/var/spool/asterisk/monitor/incoming/8452781980/In-2005110
5-002616_9142775896_PUMS IN TollFree_2781980}) in new stack
-- Executing Goto(CAPI/ISDN1/2781980-0, incoming-isdn|s|1) in
new stack
-- Goto (incoming-isdn,s,1)
-- Executing Monitor(CAPI/ISDN1/2781980-0,
wav|/var/spool/asterisk/monitor/incoming/8452781980/In-20051105-002616_
9142775896_PUMS IN TollFree_2781980}) in new stack
-- Executing GotoIfTime(CAPI/ISDN1/2781980-0,
17:31-6:59|mon-sun|*|*?2300in|s|13) in new stack
-- Goto (2300in,s,13)
-- Executing Answer(CAPI/ISDN1/2781980-0, ) in new stack
  == Spawn extension (2300in, s, 13) exited non-zero on
'CAPI/ISDN1/2781980-0'
Ouch ... error while writing audio data: : Broken pipe
Warning, flexibel rate not heavily tested!
Segmentation fault 

--
At first I thought it was me, but this is happening on two machines at
different sites.  Has anyone else experienced this?  CAPI is using ulaw.

This happens with or without the Monitor app.

On 10/28 my output is as follows:
*CLI   == ISDN1: Incoming call '19142775896' - '2781980'
-- Executing Set(CAPI/ISDN1/2781980-0, IncomingLine=2781980) in
new stack
-- Executing NoOp(CAPI/ISDN1/2781980-0, 19142775896) in new
stack
-- Executing Set(CAPI/ISDN1/2781980-0, CALLERID(name)=PUMS IN
TollFree) in new stack
-- Executing Set(CAPI/ISDN1/2781980-0, CALLERID(num)=9142775896)
in new stack
-- Executing Set(CAPI/ISDN1/2781980-0,
CALLFILENAME=/var/spool/asterisk/monitor/incoming/8452781980/In-2005110
5-003153_9142775896_PUMS IN TollFree_2781980}) in new stack
-- Executing Goto(CAPI/ISDN1/2781980-0, incoming-isdn|s|1) in
new stack
-- Goto (incoming-isdn,s,1)
-- Executing Monitor(CAPI/ISDN1/2781980-0,
wav|/var/spool/asterisk/monitor/incoming/8452781980/In-20051105-003153_
9142775896_PUMS IN TollFree_2781980}) in new stack
-- Executing GotoIfTime(CAPI/ISDN1/2781980-0,
17:31-6:59|mon-sun|*|*?2300in|s|13) in new stack
-- Goto (2300in,s,13)
-- Executing Answer(CAPI/ISDN1/2781980-0, ) in new stack
  == ISDN1: Answering for 2781980
-- Executing Wait(CAPI/ISDN1/2781980-0, 1) in new stack
  == ISDN1: Setting up echo canceller (PLCI=0x401, function=1,
options=4, tail=64)
  == ISDN1: Setting up DTMF detector (PLCI=0x401, flag=1)
-- ISDN1: Echo canceller successfully set up (PLCI=0x401)
-- Executing BackGround(CAPI/ISDN1/2781980-0, pums/thkyou) in
new stack
-- Playing 'pums/thkyou' (language 'en')
  == Spawn extension (2300in, s, 15) exited non-zero on
'CAPI/ISDN1/2781980-0'
  == ISDN1: CAPI Hangingup
CAPI INFO 0x3490: Normal call clearing  

---
As you can see, something is happening right at the Answer command.

Any insight would be appreciated. 

Regards,
Greg
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RE: [Asterisk-Users] Segfault on latest head 10/31

[gw]

Hello Rich,
I will work with this a bit more, however it only seems to happen on the
newer code.  Went up to 10/25 with no problems. 

Yes I know about module unloading, :).

Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Monday, October 31, 2005 8:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Segfault on latest head 10/31


 Anyone seen this one so far?  Seems to happen in or outgoing, and even

 if I just pick up the channel.

Nope. cvs-head from yesterday and the last several days are working just
fine on fc3. What distro are you using?

 09/15 revision works fine, but the 10/31 checkout is doing this 
 instantly.  All with HEAD zaptel and libpri
 
 Oh and another off topic thing.
 
 Sometimes I have a way of forgetting I have asterisk running, and do a

 module unload.  As you can expect, this causes an EIP and kills the 
 server.  The server will then stay stuck at the EIP, but does anyone 
 know of a way to do an auto-reboot?  Or shouldn't the zaptel channel 
 module not be unloadable while asterisk is running?  Sure I know it's 
 my fault if I do this by accident, but fortunately the server is only 
 45 mins away.  Would be rough in another state to make that mistake :)

kernel modules should not be unloaded. The fix for that is remedial
training on behalf of the sys admin (you).
 
 Asterisk Ready.
 *CLI -- Starting simple switch on 'Zap/28-1'
 Ouch ... error while writing audio data: : Broken pipe Warning, 
 flexibel rate not heavily tested!
 Segmentation fault

The above generally means you've got something very wrong in your
/etc/zaptel.conf or /etc/asterisk/zapata.conf definitions. Best guess is
the zap channels defined in zaptel.conf don't match those defined in
zapata.conf.


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RE: [Asterisk-Users] Segfault on latest head 10/31

[gw]

I am on Debian testing.

Using this script:
rmmod wctdm
rmmod wcfxo
rmmod wct4xxp
rmmod zaptel

modprobe zaptel
modprobe wct4xxp
modprobe wcfxo
modprobe wctdm 

wctdm  41536  0
wcfxo  13344  0
wct4xxp   104128  30
zaptel163844  73 wctdm,wcfxo,wct4xxp  

Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Monday, October 31, 2005 9:10 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Segfault on latest head 10/31

On Monday 31 October 2005 05:11, [EMAIL PROTECTED] wrote:
 Sometimes I have a way of forgetting I have asterisk running, and do a

 module unload.  As you can expect, this causes an EIP and kills the 
 server.  The server will then stay stuck at the EIP, but does anyone 
 know of a way to do an auto-reboot?  Or shouldn't the zaptel channel 
 module not be unloadable while asterisk is running?  Sure I know it's 
 my fault if I do this by accident, but fortunately the server is only 
 45 mins away.  Would be rough in another state to make that mistake :)

...??  

[EMAIL PROTECTED]:~# rmmod wct4xxp
ERROR: Module wct4xxp is in use
[EMAIL PROTECTED]:~# rmmod zaptel
ERROR: Module zaptel is in use by wct4xxp

I can't remove them when Asterisk is running.

What distro?  lsmod should show a nonzero use count for your zaptel
and lowlevel hardware driver.

-A.
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[Asterisk-Users] Segfault on latest head 10/31

[gw]

Anyone seen this one so far?  Seems to happen in or outgoing, and even
if I just pick up the channel.

09/15 revision works fine, but the 10/31 checkout is doing this
instantly.  All with HEAD zaptel and libpri

Oh and another off topic thing.

Sometimes I have a way of forgetting I have asterisk running, and do a
module unload.  As you can expect, this causes an EIP and kills the
server.  The server will then stay stuck at the EIP, but does anyone
know of a way to do an auto-reboot?  Or shouldn't the zaptel channel
module not be unloadable while asterisk is running?  Sure I know it's my
fault if I do this by accident, but fortunately the server is only 45
mins away.  Would be rough in another state to make that mistake :)

Asterisk Ready.
*CLI -- Starting simple switch on 'Zap/28-1'
Ouch ... error while writing audio data: : Broken pipe
Warning, flexibel rate not heavily tested!
Segmentation fault



Greg
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RE: [Asterisk-Users] Echo Canceller question- is there a viablesolution?

[gw]

Hello Matthew,
It is always nice to see improvements.  I look forward to testing your
patches.

It just seems that so many other hardware manufacturers have tackled the
problem, I am surprised digium has not put more research into getting
the issue solved in software, which is possible, as opposed to coming up
with alternate solutions.

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Fredrickson
Sent: Thursday, October 27, 2005 1:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Echo Canceller question- is there a
viablesolution?


On Oct 27, 2005, at 12:38 AM, [EMAIL PROTECTED] wrote:
 My question is, what is the direction in relation to analog boards and

 such?

Right now, it looks like the current fad of the asterisk group is
hardware echo cancelation.  However, there is work that is occurring on
the software echo cans to improve them.  In fact, I just committed
basically an update to
KB1
(which was until now the latest and greatest version of MEC2) that is
supposed to provide somewhat significant improvements.


 Quite a few people tend to have difficulties with echo, and although 
 the WIKI has some very helpful advice, from a business standpoint I 
 would think that it would be an important step to come up with a final

 solution to the problem.

 Many companies who make the higher end equipment seem to have tackled 
 the issue on their hardware.

 Do we know if digium is spending time on solving the issue?  For 
 example, having a tool to run on a digium analog or t1 board to 
 analyze the line statistics and come up with the proper gain settings 
 could be extremely helpful.

 Such a tool would require a firm knowledge of the causes and solutions

 to echo however, but I would assume that digium should have a grasp on

 this.

 It just seems difficult to suggest to companies to use an asterisk 
 based solution (if they do not use pri) when there is the possibility 
 that an installation will have issues with echo.

 At this point, it feels more like a trial experience to eliminate echo

 in various environments.

Unfortunately, that's the way it is right now.  Getting to the point
where you have enough knowledge to be able to work on these things is
not an insignificant task.
It seems like we're slowly getting there, and now that we have some more
interest on improving the software echo cans we might be a little be
closer to getting to the point where it just works.


 I have used local tone from the CO to help narrow things down, but a 
 tool that would loop dial a line and do an analysis could reduce the 
 implementation time from days to hours.

Well, there isn't anything that does the whole job right now.  
There's a bunch
of pieces that go together, and if you have the necessary knowledge of
how to put the pieces together, you can get pretty close to it just
working. 
  It's not that
bad though, one can also see it as job security as well :-)

Matthew Fredrickson

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RE: [Asterisk-Users] [Sorta OT] Eicon DIVA with [EMAIL PROTECTED]

[gw]

This is a DIVA Server card correct?  Regular diva or diva pro will not
work as far as I know.

Regards,
Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, October 26, 2005 4:31 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] [Sorta OT] Eicon DIVA with [EMAIL PROTECTED]

Hi,

I have an AAH installation with an active Eicon DIVA BRI card. My AAH is
built on Centos 3.5 which is at kernel
2.4.21.37.EL.

I have installed the source level RPM from Eicon as well as
chan_capi-0.3.5.

When I try to run divactrl load -c 1 -f ETSI -Debug I get a response:
A: can't get card type for DIVA adapter number 1

I have been reading and following the instructions and advice in the
following two wiki's;

http://www.voip-info.org/wiki/index.php?page=Asterisk+How+to+connect+wit
h+CAPI
http://www.voip-info.org/wiki/index.php?page=Asterisk+Eicon+Diva+CAPI+IS
DN

I am going around in circles with no joy.

Does anyone know of a clear concise guide to get this card to work with
AAH.

Uwin

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[Asterisk-Users] Echo Canceller question- is there a viable solution?

[gw]

Hello All,

I have a question in regards to the echo cancellation mechanism's used
in HEAD.

I know a lot of effort has been made to reduce echo on the various
boards, and Digium has introduced the new echo cancelling boards.

My question is, what is the direction in relation to analog boards and
such?

Quite a few people tend to have difficulties with echo, and although the
WIKI has some very helpful advice, from a business standpoint I would
think that it would be an important step to come up with a final
solution to the problem.

Many companies who make the higher end equipment seem to have tackled
the issue on their hardware.

Do we know if digium is spending time on solving the issue?  For
example, having a tool to run on a digium analog or t1 board to analyze
the line statistics and come up with the proper gain settings could be
extremely helpful.

Such a tool would require a firm knowledge of the causes and solutions
to echo however, but I would assume that digium should have a grasp on
this.

It just seems difficult to suggest to companies to use an asterisk based
solution (if they do not use pri) when there is the possibility that an
installation will have issues with echo.

At this point, it feels more like a trial experience to eliminate echo
in various environments.

I have used local tone from the CO to help narrow things down, but a
tool that would loop dial a line and do an analysis could reduce the
implementation time from days to hours.

I have clients which I would jump on if I could just go to their site,
do an install, and not have to worry about these kinds of issues (for
example, if I fly out for a day to do an install, it would be a big
problem to have to make multiple trips to solve an echo issue)

I personally have had to deal with echo problems as well, but been able
to manage.  None of the solutions I have come up with however have left
me feeling 100 percent satisfied.

I am talking for the most part on the local end, using a hybrid of local
pbx's and land lines, and T1.

Any ideas?

Regards,
Greg
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RE: [Asterisk-Users] Terrible echo with Te110P and Adit 600

[gw]

Yes I did notice it immediately.  I intend to tweak more, but for the
moment it seems like echo is minimized to zero.

This is a big step up from where I was.  Now I just need to see if it
bothers people at the office.

Also been looking for a way to restore CNG (comfort noise) to avoid the
'are you there' issues.  No luck on researching it with t1 yet though.

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, October 24, 2005 3:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Terrible echo with Te110P and Adit 600



On Sun, 23 Oct 2005, C F wrote:

 Why?
 
 On 10/23/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
  On Sunday 23 October 2005 18:02, C F wrote:
   Sorry guys I forgot to mention that in my setup I always enable 
   agressive in zconfig
 
  Yuck.  I find the agressive echo canceller totally unacceptable.


Did you listen to the aggressive suppressor working?  Every time you
speak, the other end of the line gets muted dead.  

I guess if you have to use it then you have to use it.  But I wouldn't
make it my default.

Steve

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RE: [Asterisk-Users] Terrible echo with Te110P and Adit 600

[gw]

Followup, I set a -2.0 gain from my asterisk t1  pbx, and echo seems
mostly gone.

A note, I also turned on the aggressive suppressor in zconfig.h

Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gregory
Wiktor - ADCom Corp.
Sent: Saturday, October 22, 2005 11:07 AM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Terrible echo with Te110P and Adit 600

I have a similar setup... I set the canceller on the incoming PSTN
lines, but turn it off on the FXS.  

I have no local internal echo over the t1, but moderate over the PSTN.
I managed to tweak it a little and most of my  outbound (local side)
echo is minimized, but still there a little. I have no incoming echo.

You mind elaborating on where you are getting the echo? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren
Wright
Sent: Tuesday, October 18, 2005 10:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Terrible echo with Te110P and Adit 600


8 POTS lines - Adit 600 - TE110P - Dell Precision 530 Dual Xeon running
1.0.9 and 1.2 (tried both)


The echo is insurmountable.  I have tried everything, and the pots lines
are clean.  If I go from an FXO on the Adit 600 straight to an FXS, I
get no echo from an analog phone.  

I put an 128ms T1 echo canceller in between the adit and the TE110P, and
the echo was still horrible.  

I finally disabled the Zapata echo cancellerand WHAMMO!  It's
perfect now.  

The TE110P is on it's own IRQ.. and the machine has PLENTY of
horsepower.

Any ideas so I don't have to spend $1000 on an echo canceller?

-Darren




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RE: [Asterisk-Users] Callerid on t1 lines

[gw]

I had this, my problem turned out to be in zapata.conf on the receiving
end.

I'll do the KS, right now I am using LS.  Any particular reason to use
KS?  The LSCPD on the adit seems to work fairly decently. 

Now I just need to work out some echo, although I have done milliwatt
tests to a local line, I still seem to get echo at the beginning of a
call regardless of how I set the training.

Thanks,
Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Monday, October 17, 2005 1:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Callerid on t1 lines

How are you checking if CallerID is received?
You should do at least a Noop(${CALLERIDNUM}) or if running head:
Noop(${CALLERID(NUM)}) so that you can verify that.
How do you know that your telco is giving you CID?
If you live in the US then setup the Adit to do LSCPD and Asteisk as
ks_fxs. and not loop start.

On 10/17/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
 Hello,
 That's what I really needed to know, that it was possible.

 Here is my setup:

 Telco Analog W/CID  FXO ADIT600 LoopStart  Loopstart Asterisk T1.

 Then LoopStart Asterisk T1  Loopstart Panasonic DBS PBX T1.

 At this point, I do not see any CID coming in from the telco into 
 asterisk.  Even when I increase the wait time, and the zapata.conf has

 asreceived set.

 I tried EM from the dbs to asterisk, but would get no dialtone from 
 asterisk as it was not working properly with immediate mode.

 The main purpose of the setup is to do call recording on 3 analog and 
 2 bri lines, and pass them to the pbx transparently.  Also to allow * 
 transfers and queuing.

 Thanks,
 Greg

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of C F
 Sent: Saturday, October 15, 2005 9:16 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Callerid on t1 lines

 What is the adit 600 doing? FXO? FXS? how you connected to the PSTN?
 I got an Adit 600 with both FXO and FXS as well as a PRI and I'm 
 getting CallerID on all three.

 On 10/14/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
  Hello All,
  Just a question, I have an adit600 and I am looking for a way to 
  pull the incoming cid into asterisk.
 
  Does anyone know if this is just not possible via t1? Or is it only 
  available on PRI?
 
  Thanks,
  Greg
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RE: [Asterisk-Users] Terrible echo with Te110P and Adit 600

[gw]

I have a similar setup... I set the canceller on the incoming PSTN
lines, but turn it off on the FXS.  

I have no local internal echo over the t1, but moderate over the PSTN.
I managed to tweak it a little and most of my  outbound (local side)
echo is minimized, but still there a little. I have no incoming echo.

You mind elaborating on where you are getting the echo? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren
Wright
Sent: Tuesday, October 18, 2005 10:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Terrible echo with Te110P and Adit 600


8 POTS lines - Adit 600 - TE110P - Dell Precision 530 Dual Xeon running
1.0.9 and 1.2 (tried both)


The echo is insurmountable.  I have tried everything, and the pots lines
are clean.  If I go from an FXO on the Adit 600 straight to an FXS, I
get no echo from an analog phone.  

I put an 128ms T1 echo canceller in between the adit and the TE110P, and
the echo was still horrible.  

I finally disabled the Zapata echo cancellerand WHAMMO!  It's
perfect now.  

The TE110P is on it's own IRQ.. and the machine has PLENTY of
horsepower.

Any ideas so I don't have to spend $1000 on an echo canceller?

-Darren




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RE: [Asterisk-Users] Callerid on t1 lines

[gw]

Hello,
That's what I really needed to know, that it was possible.

Here is my setup:

Telco Analog W/CID  FXO ADIT600 LoopStart  Loopstart Asterisk T1.

Then LoopStart Asterisk T1  Loopstart Panasonic DBS PBX T1.

At this point, I do not see any CID coming in from the telco into
asterisk.  Even when I increase the wait time, and the zapata.conf has
asreceived set.

I tried EM from the dbs to asterisk, but would get no dialtone from
asterisk as it was not working properly with immediate mode.

The main purpose of the setup is to do call recording on 3 analog and 2
bri lines, and pass them to the pbx transparently.  Also to allow *
transfers and queuing.

Thanks,
Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C F
Sent: Saturday, October 15, 2005 9:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Callerid on t1 lines

What is the adit 600 doing? FXO? FXS? how you connected to the PSTN?
I got an Adit 600 with both FXO and FXS as well as a PRI and I'm getting
CallerID on all three.

On 10/14/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
 Hello All,
 Just a question, I have an adit600 and I am looking for a way to pull 
 the incoming cid into asterisk.

 Does anyone know if this is just not possible via t1? Or is it only 
 available on PRI?

 Thanks,
 Greg
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[Asterisk-Users] Callerid on t1 lines

[gw]

Hello All,
Just a question, I have an adit600 and I am looking for a way to pull
the incoming cid into asterisk.

Does anyone know if this is just not possible via t1? Or is it only
available on PRI?

Thanks,
Greg
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RE: [Asterisk-Users] Asterisk PBX in Debian

[gw]

Why bother with packages anyhow?  I just installed debian base and did a
cvs get for head, and all good to go.

Besides I found that using packages with asterisk on debian can do odd
things to your custom sound files if you do a remove.

Regards,
Greg

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Friday, October 07, 2005 7:17 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Asterisk PBX in Debian

On Fri, Oct 07, 2005 at 12:51:25AM -0500, Carlos Prieto wrote:
 Hi everyone.
 
 I've installed Asterisk PBX using apt packages, but i don't have 
 actually any Digium card, so i want to use ztdummy.
 
 I've tried to modify the Makefiles in the debian source package, i 
 don't get any error, but still the ztdummy module doesn't get
compiled.

What file exactly did you edit? What command did you run? I suspect the
file you have edited got overrun.

 
 Does anybody has idea how to get the ztdummy module using the debian 
 package system?

I'm not sure you need. Packages from 

  deb http://rapid.dotsrc.org/rapid sarge main

already have ztdummy and ztdummy is on by default in the zaptel-source
package.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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RE: [Asterisk-Users] Asterisk PBX in Debian

[gw]

 -Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Friday, October 07, 2005 8:17 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Asterisk PBX in Debian

On Fri, Oct 07, 2005 at 07:37:26AM -0400, [EMAIL PROTECTED] wrote:
 Why bother with packages anyhow?  I just installed debian base and did

 a cvs get for head, and all good to go.

And if you have several systems?

I would make a custom package in that case, for easy updating.  Depends
of course if you are using head or not.

 
 Besides I found that using packages with asterisk on debian can do odd

 things to your custom sound files if you do a remove.

Regarding the sounds files: I don't think that the way Asterisk
installer handles them is very optimal either.

Your message got me thinking, though. I believe that Debian is right
installing all sounds to /usr/share/asterisk/sounds . But
/var/lib/asterisk/sounds (/usr/local/asterisk/sounds? ) should still be
kept for custom sounds that are never touched by the package. 

I figure that file.c:build_filename could be changed to do the
following:

  if exists /var/lib/asterisk/sounds/filename
return /var/lib/asterisk/sounds/filename
  else if exists /usr/share/sounds/asterisk/filename
return /usr/share/sounds/asterisk/filename

What do you think? I figure I'll try to push this into Debian first.
(If this is indeed a good idea)

Using /var works, but setting it in asterisk could be a pain when it
comes to voicemail prompts.  Plus, extensions.conf would need to grow
and become a little cluttered.  Unless of course, one could do something
to specify a new root voicemail path, and if the file is not found it
plays from the default.

Greg
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RE: [Asterisk-Users] Snom 360 Phones - Administrator/User Feedback

[gw]

Well, The Polycom  Cisco are high-end. I have 
used others like a pingtel, sipura 841, etc. Nothing has the 'feel' of the 
cisco, and nothing has the functionality of the polycom (like call drop from 
conferences). Next I will try the aastra wireless combo phone for the 
office. Looks nice what it offers, but I don't expect it to be as nice as 
the cisco/polycoms.

It's all plus and minus depending on the use really. If you 
are on the phone all day on a headset, perhaps I would choose cisco. If 
you do a lot of multiple calls and need to add/drop people, or if you need BLF, 
go with the polycoms. It was worth the money for me no 
doubt.

Greg


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Omar A. 
SabekSent: Thursday, October 06, 2005 7:37 PMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: 
[Asterisk-Users] Snom 360 Phones - Administrator/User 
Feedback
I'm looking for some feedback on the Snom 360 phone. After deploying 
six of them to an office, I'm not as enthusiastic about them as I was when I was 
testing one before the deployment. The firmware seems to be consistently buggy, 
some of its problems are intermittent which makes it frustrating to troubleshoot 
and the support from Snom is lackluster to say the least. I find myself favoring 
Cisco and Polycom phones, not only from a user POV but also the automation of 
deployment... Does anyone share similar sentiments?Omar 
Sabek
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RE: [Asterisk-Users] Sprint Nextel sueing over VoIP patents

[gw]

They probably have liability insurance anyhow, plus this is all too
often standard business practice.

Will it hold up? Who knows, someone will probably go under from the
legal fees alone.

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Riddell
Sent: Wednesday, October 05, 2005 6:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Sprint Nextel sueing over VoIP patents

trixter http://www.0xdecafbad.com wrote:
 Anyone thinking about doing a VoIP business may want to get more info 
 before proceeding since they may not have the millinos vonage has to 
 fight this.

Unless of course they don't live in the United Sue'ers of America.

:D

--
Cheers,

Matt Riddell
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RE: [Asterisk-Users] Monitor and sox mix quality

[gw]

This is what I ended up doing...Using sox to mix then lame to encode.
Ends up with a much larger audio file though. Not a problem, but since
the channels are split it sounds much better as they don't get
artifacts.

Greg

#!/usr/bin/perl
@lines=`ls -1 /var/spool/asterisk/monitor/incoming/99/*in.wav`;
foreach $line (@lines){
(@splits)=split(in.wav,$line);
chop(@splits[0]);
[EMAIL PROTECTED];
$base=~ s/()/\\\/g;
$base=~ s/( )/\\\ /g;
$base=~ s/()/\\\/g;
$base=~ s/()/\\\/g;
print `/usr/bin/nice -n 20 /usr/local/bin/sox $base-in.wav -c 2
$base-in-l.wav pan -1`;
print `/usr/bin/nice -n 20 /usr/local/bin/sox $base-out.wav -c 2
$base-out-r.wav pan 1`;
print `/usr/bin/nice -n 20 /usr/local/bin/soxmix $base-in-l.wav
$base-out-r.wav $base.wav`;
print `/usr/bin/nice -n 20 /usr/bin/lame -S --cbr -b32 -m s
$base.wav $base.mp3`;
`rm -f $base.wav`;
`rm -f $base-*`;} 
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
Feally
Sent: Monday, September 19, 2005 2:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Monitor and sox mix quality

I have not noticed any issues with quality, just with caller volumes
being way different when mixing 2 channel types (ZAP and SIP
specifically). Here's my custom script for processing the recording
files. Make sure you use option m on your monitor command so that the
custom script will run. My script makes a stereo mp3 with the 2 people
split to left/right and makes the 2 sides have an equal max volume. Hope
this helps. You can always modify the script to adjust how the mp3 is
encoded.


extensions.conf:

[globals]
MONITOR_EXEC=/usr/local/bin/2wav2mp3

[macro-callext]
s,1,monitor(wav|${ARG1}_${TIMESTAMP}|m)
s,2,dial(SIP/${ARG1})


[EMAIL PROTECTED]:/usr/local/bin# cat /usr/local/bin/2wav2mp3 #!/bin/sh #
2wav2mp3 - create stereo mp3 out of two mono wav-files # source files
will be deleted # # usage: 2wav2mp3 wave1 wave2 mp3 # #
extensions.conf # use option m on monitor command # add this variable
to [globals] # MONITOR_EXEC=/usr/local/bin/2wav2mp3


# location of SOX and SOXMIX
# (set according to your system settings, eg. /usr/bin) SOX=nice -n 20
/usr/bin/sox
SOXMIX=nice -n 20 /usr/bin/soxmix
LAME=nice -n 20 /usr/local/bin/lame -S --cbr -b32 -m s
NORMALIZE=nice -n 20 /usr/bin/normalize --no-progress -a 1.0 --peak

# command line variables
LEFT=`echo $1 | awk -F.wav '{print $1}'` RIGHT=`echo $2 | awk -F.wav
'{print $1}'` OUT=`echo $3 | awk -F.wav '{print $1}'`

#test if input files exist
test ! -r $LEFT.wav  exit
test ! -r $RIGHT.wav  exit

# convert mono to stereo, adjust balance to -1/1 $NORMALIZE $LEFT.wav
$NORMALIZE $RIGHT.wav # left channel $SOX $LEFT.wav -c 2 $LEFT-tmp.wav
pan -1 # right channel $SOX $RIGHT.wav -c 2 $RIGHT-tmp.wav pan 1

# in case an old version of sox is used, encoding # can be done
afterwards $SOXMIX -v 1 $LEFT-tmp.wav -v 1 $RIGHT-tmp.wav -v 1 $OUT.wav
$LAME $OUT.wav $OUT.mp3


#remove temporary files
test -w $LEFT-tmp.wav  rm $LEFT-tmp.wav test -w $RIGHT-tmp.wav  rm
$RIGHT-tmp.wav test -w $OUT.wav  rm $OUT.wav

#remove input files if successfull
test -r $OUT.mp3  rm $LEFT.wav $RIGHT.wav # eof


Good Luck!

-Jon


[EMAIL PROTECTED] wrote:

Hello All,
I am using monitor with soxmix, however the quality seems somewhat low 
after sox converts to mp3.

Does anyone know a way to get a higher quality file?  Some of my lines 
are coming in on isdn.

Regards,
Greg
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RE: [Asterisk-Users] Real Life FAX sending receiving

[gw]

I can send/receive just fine on an eicon bri to a zaptel analog
interface.

I would say, if you wish to use faxing on a regular basis to a remote
proxy though, you're possibly better off with a landline.

Regards,
Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Monday, October 03, 2005 3:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Real Life FAX sending receiving

Jenna Cole wrote:

receive the fax via SIP and send it to my faxmachine.
I also want to send a fax from my faxmachine through the digium card, 
so asterisk should send the fax via SIP to the gateway, which also has 
a faxmachine connected.

is this possible?
  

Short answer, no.  Long answer can be found here:

http://www.soft-switch.org/spandsp_faq/ar01s04.html

Doug

-- 
 
Ben Franklin quote:

Those who give up essential liberties for temporary safety deserve
neither liberty nor safety.


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RE: [Asterisk-Users] Revieving some fax problems

[gw]

I had some trouble going from brizaptel analog, but once I got the gain
settings right, I would say it has worked well. Don't have stats, but
any faxes I do send to it seem to go through.

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph
Sent: Friday, September 30, 2005 4:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Revieving some fax problems

Try Hylafax, with external fax/modem, it works 99.999% It you try to
route it via Asterisk (with NVFaxDetect) your success will be about 95%

--
#Joseph

On Fri, 2005-09-30 at 15:57 -0400, Alexandre Leclerc wrote:
 Hi,
 
 We are recieving some faxes, but I would say that about 50% of them do

 not work. We don't know why... is it something with the faxes speed, 
 volume, etc? Should we use a real fax machine?
 
 Using a TDM13B with a rxgain of about 5.0...
 
 Thank you for any help.
 

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RE: [Asterisk-Users] Diva

[gw]

Nope. At least I tried and never could get it 
working. It's a semiactive.


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Giordano 
GrandisSent: Friday, September 30, 2005 6:59 AMTo: 
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: 
[Asterisk-Users] Diva


Hi all,
just a question: can i 
use this kind of diva for asterisk?

00:14.0 Network controller: Eicon 
Networks Corporation Diva ISDN Pro 3.0 PCI

Thanks 
all

Giordano 


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[Asterisk-Users] Voice Prompts, what do you think? Good voice. Should we record a new prompt-set?

[gw]

Hello all,
I have someone working for me who has a nice phone voice.  I looked at
some available prompts for asterisk, and found both the free and
commercial ones to be pretty horrible.  The asterisk ones are good, but
I wish I had more to choose from sometimes.

My question is, what do you think, should I bother having her record a
full spectrum of prompts for asterisk?

If you want to hear her voice, the ivr/bri number is 1-914-693-0821

If it is something the community would want, I can spend the time with
her, otherwise I am fine with the regular voices.

If you wanted any custom prompts done, we could do it.  She works for me
anyway so its not a big deal, but for anything extensive I should give
her some money to do it.

This is not a studio setup, but we have a good microphone setup and the
quality even on BRI sounds great. I just need to spend some time
adjusting the audio levels on my bri and in the wav files.

For me, this worked out well, since my own voice I admit sounds very
anal on the phone.

Feel free to call and listen, but the timeout goes to fax.

There is a good chance I will do it, but want some feedback. What would
be especially helpful at this point would be suggestions for new prompts
for asterisk.  This way if I spend like 2 hours doing it with her, at
least I can cover any new prompts that may come up.

I like the Allison prompts, but sometimes they sound too sexy.  April's
voice(my employee) is a bit flirty, but I think it works in a good way.

If I did it I would probably give away a basic prompt set, but charge
$50 or $100 for the extra stuff like pin codes and things that are not
normally used for a personal nature.

Regards,
Gregory Wiktor
[EMAIL PROTECTED]
Web: www.adcomcorp.com
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[Asterisk-Users] Voice Prompts, what do you think? Good voice.

[gw]

Repost, first one never made it to the list... 

-Original Message-
From: Gregory Wiktor - ADCom Corp. 
Sent: Thursday, September 29, 2005 3:20 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Voice Prompts, what do you think? Good voice. Should we record
a new prompt-set?

Hello all,
I have someone working for me who has a nice phone voice.  I looked at
some available prompts for asterisk, and found both the free and
commercial ones to be pretty horrible.  The asterisk ones are good, but
I wish I had more to choose from sometimes.

My question is, what do you think, should I bother having her record a
full spectrum of prompts for asterisk?

If you want to hear her voice, the ivr/bri number is 1-914-693-0821

If it is something the community would want, I can spend the time with
her, otherwise I am fine with the regular voices.

If you wanted any custom prompts done, we could do it.  She works for me
anyway so its not a big deal, but for anything extensive I should give
her some money to do it.

This is not a studio setup, but we have a good microphone setup and the
quality even on BRI sounds great. I just need to spend some time
adjusting the audio levels on my bri and in the wav files.

For me, this worked out well, since my own voice I admit sounds very
anal on the phone.

Feel free to call and listen, but the timeout goes to fax.

There is a good chance I will do it, but want some feedback. What would
be especially helpful at this point would be suggestions for new prompts
for asterisk.  This way if I spend like 2 hours doing it with her, at
least I can cover any new prompts that may come up.

I like the Allison prompts, but sometimes they sound too sexy.  April's
voice(my employee) is a bit flirty, but I think it works in a good way.

If I did it I would probably give away a basic prompt set, but charge
$50 or $100 for the extra stuff like pin codes and things that are not
normally used for a personal nature.

Regards,
Gregory Wiktor
[EMAIL PROTECTED]
Web: www.adcomcorp.com
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RE: [Asterisk-Users] chan_capi-cm, Euro ISDN bus: 2 extensions on same BRI port not working

[gw]

Don't know, but at least check your capi.conf.  I had a similar issue,
but from the telco  asterisk.

Regards,
Greg

[general]
nationalprefix=1
internationalprefix=011
rxgain=0.5
txgain=0.5
ulaw=yes;set this, if you live in u-law world instead of a-law
 ;(2 makes sense for single BRI, 30 for PRI)

[ISDNL1]  ;this example interface gets name 'ISDN1' and may be
any;name not starting with 'g' or $;ntmode=yes  ;if isdn card
operates in nt mode, set this to yes
isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward
dial);when using NT-mode, ptp shoul$;isdnmode=did ;'MSN'
(point-to-multipoint) or 'DID' (direct inward dial);when using NT-mode,
ptp shou$;incomingmsn=*;allow incoming calls to this list of
MSNs/DIDs, * == any
incomingmsn=5912211;allow incoming calls to this list of MSNs/DIDs,
* == any
controller=1 ;capi controller number to use
group=1  ;dialout group
;prefix=0;set a prefix to calling number on incoming calls
;softdtmf=on  ;enable/disable software dtmf detection, recommended
for AVM cards
;relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf
detection
;accountcode= ;Asterisk accountcode to use in CDRs
context=capi-in-5912211  ;context for incoming calls
holdtype=hold;when Asterisk puts the call on hold, ISDN HOLD will be
used. If
 ;set to 'local' (default value), no hold is done and
Asterisk may
 ;play MOH.
;immediate=yes   ;immediate start of pbx with extension 's' if no digits
were
 ;received on incoming call (no destination number yet)
;echosquelch=1   ;_VERY_PRIMITIVE_ echo suppression
echocancel=yes
;echocancel=yes  ;EICON DIVA SERVER (CAPI) echo cancelation
 ;(possible values: 'no', 'yes', 'force', 'g164',
'g165')
;echocancelold=yes;use facility selector 6 instead of correct 8
(necessary for older eicon drivers)
echotail=64 ;echo cancel tail setting
bridge=yes  ;native bridging (CAPI line interconnect) if available
callgroup=1 ;Asterisk call group
;deflect=8161000 ;deflect incoming calls to 1234567 if all B channels
are busy
devices=1;number of concurrent calls on this controller

[ISDNL2]
isdnmode=msn
incomingmsn=6930821
controller=1
callgroup=2
group=2
bridge=yes  ;native bridging (CAPI line interconnect) if available
context=capi-in-6930821
echocancel=no
devices=1 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, September 28, 2005 8:36 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] chan_capi-cm,Euro ISDN bus: 2 extensions on
same BRI port not working

Hello,

I am using a system with an AVM ISDN PCI card (fcpci) and asterisk with
chan_capi-cm-0.6. The hardware is connected to a Siemens Hipath 3550
PBX. As a BRI connection has 2 channels and allows 2 simultaneous calls,
numbers/MSNs 6391 and 6392 were for provisioned for each channel. The
system is working (partly, read on), the trick is the correct cable
wiring and setup the PBX's port as S0 Euro Bus.

Calls from asterisk to PBX are working ok, really nice! But calls from
PBX to asterisk are partly working. Calls to 6391 are okay, but calls to
6392 are not. They simply did not appear on the BRI port (checked with
capi debug).

The problem seems to be something related to MSN routing. The PBX manual
says it is possible to connect up to 64 devices in one S0/BRI port and
even show the wiring diagram for that. However, there is not a clear way
to define MSN x goes to channel 1, MSN y goes to channel 2 in the PBX
management software. I am starting to think it is impossible to
configure this through PBX...

The PBX manual and on-line help says (as far as I understood) MSNs are
configured at the user station but, this time, I was unable to
configure MSN x=channel 1, MSN y=channel 2 in the capi.conf file of
chan_capi which acts as the station.

Actually, I am super-confused! Is it possible to configure the MSN
routing as I explained? Is Siemens right, that is, it is up to the
station to advertise it is reachable and its MSN? Will chan_capi +
fcpci do that?

Thanks in advance,

--hg


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RE: [Asterisk-Users] BRI Hunting, using both channels on one msn

[gw]

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
Sent: Saturday, September 24, 2005 2:30 PM
To: Gregory Wiktor - ADCom Corp.
Cc: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] BRI Hunting, using both channels on one
msn

On Fri, 23 Sep 2005 [EMAIL PROTECTED] wrote:
 Hello Armin,
 I tried your new version of chan capi and it works well.
 
 I did have one question about capi.conf.  I have a bri with 2 spids, 
 but I want to have the second go to a zap fax channel.
 
 Right now I can direct it, but the echo canceller is setting up.  Do 
 you know a way to cancel it?  Fax works, but I suspect it would work 
 better with EC off on large faxes.

I don't know enough about the spid stuff, but you should be able to
create two interfaces in capi.conf (instead of one), which devices=1 for
each.
So you should be able to set different settings for each channel.

Ok, I managed to setup two but had devices=1

When using fax via capi with Eicon cards, the echo canceler should not
be used automatically.

This is correct, however when I do it on an analog machine connected to
a zap channel is what I am concerned about.

I managed to also get a local modem to work via the zaptel port, and
connect at 50kps, although asterisk kept mentioning a fax detect while
the modem would connect.

 I tried the capi fax receive, but the images came out with the wrong 
 dimensions(on .05).

Maybe a problem with setting fine/normal resolution?

Turned out to be photoshop, another viewer worked ok.
 
 Also,
 Is there a way to split each msn into a different call group in 
 capi.conf?  I tried a few combinations but no luck.
 I was thinking I could disable the EC for the line in general.

See above.

 Oh as per the hunt, I had verizon program a hunt into the line and it 
 seems to work now.  It is funny though, since I think my usrobotics 
 modem can also do it, I just don't know exactly how it is handled.  
 capi just causes the line to report a busy.
 
 And there is a new eicon driver that works on 2.6.

Which one do you mean?

The new driver? Came out early sept.  Now the scripts work on debian and
you don't need to manually initialize it.

Armin
 
 Thanks,
 Greg
 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
 Sent: Tuesday, August 16, 2005 4:49 AM
 To: Gregory Wiktor - ADCom Corp.
 Subject: RE: [Asterisk-Users] BRI Hunting, using both channels on one 
 msn
 
 Hi Greg,
 
 now I understand. You use NI-1 with spids. I'm sorry, I don't know 
 anything about this protocol. ETSI does not have this
'channel-problem'.
 
 Maybe it can be solved with some load parameters for the BRI card.
 You should use the latest driver and divactrl (possibly the SRPM from 
 Eicon).
 
 regards,
 Armin
 
 On Tue, 16 Aug 2005 [EMAIL PROTECTED] wrote:
   Hello Armin,
  My setup is as follows:  I have 1 bri with 2 spid's, or msn's.  
  2781980 and 2781984.
  
  If a call comes in to 2781980, and is active, and another call comes

  in to 2781980, the second call will be busy.
  
  A call to 278-1984 will proceed while the 1980 is busy.
  
  The telco tells me though that the bri should be capable of hunting 
  on
 
  it's own.
  
  I did this in the past with modem banks, but they were on top of 
  centrex.
  
  What I would like to do is put an 800 number to point to the 
  278-1980,
 
  and for the most part not use the 278-1984 except for maybe a disa.
  
  The eiconctrl monitor app is aware that the line is busy, and I do 
  not
 
  believe it is notifying asterisk of the issue.
  
  I am trying to move some lines to bri since my audio quality on pots

  has been horrible.  The isdn is great, especially since you told me 
  of
 
  the ulaw modification I needed to make...  I got lucky with this 
  one, since they really could not install it without doing special 
  construction, which I managed to avoid paying the big bucks for 
  because the csr was nice about the 3 month delay.  I set it up 
  through
 
  a panasonic dbs so the secretary can just hit a button, and I get 
  immediate rings on 4 sip phones and my cell.  I would love a PRI, 
  but only need 4 channels max which is why I went with the bri.
  
  Compared to pots, the isdn is way better.  I also find it much more 
  stable than IP, to the point where it is worth the 1c/minute to use.
  
  Thanks for the help.
  
  Greg
  
  -Original Message-
  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
  Sent: Tuesday, August 16, 2005 2:06 AM
  To: Gregory Wiktor - ADCom Corp.
  Subject: RE: [Asterisk-Users] BRI Hunting, using both channels on 
  one msn
  
  On ISDN, the second channel is automatically used if the first 
  channel
 
  is busy.
  Normaly you never get a busy signal, just because ONE channel is
busy.
  Only if there is no application/phone available for that MSN, then 
  you
 
  get busy.
  Or maybe I just don't understand what you are doing...
  
  Armin
  
  On Mon, 15 Aug 2005 [EMAIL PROTECTED] wrote:
   

RE: [Asterisk-Users] BRI Hunting, using both channels on one msn

[gw]

Hello Armin,
I tried your new version of chan capi and it works well.

I did have one question about capi.conf.  I have a bri with 2 spids, but
I want to have the second go to a zap fax channel.

Right now I can direct it, but the echo canceller is setting up.  Do you
know a way to cancel it?  Fax works, but I suspect it would work better
with EC off on large faxes.
I tried the capi fax receive, but the images came out with the wrong
dimensions(on .05).

Also,
Is there a way to split each msn into a different call group in
capi.conf?  I tried a few combinations but no luck.
I was thinking I could disable the EC for the line in general.

Oh as per the hunt, I had verizon program a hunt into the line and it
seems to work now.  It is funny though, since I think my usrobotics
modem can also do it, I just don't know exactly how it is handled.  capi
just causes the line to report a busy.

And there is a new eicon driver that works on 2.6.


Thanks,
Greg
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, August 16, 2005 4:49 AM
To: Gregory Wiktor - ADCom Corp.
Subject: RE: [Asterisk-Users] BRI Hunting, using both channels on one
msn

Hi Greg,

now I understand. You use NI-1 with spids. I'm sorry, I don't know
anything about this protocol. ETSI does not have this 'channel-problem'.

Maybe it can be solved with some load parameters for the BRI card.
You should use the latest driver and divactrl (possibly the SRPM from
Eicon).

regards,
Armin

On Tue, 16 Aug 2005 [EMAIL PROTECTED] wrote:
  Hello Armin,
 My setup is as follows:  I have 1 bri with 2 spid's, or msn's.  
 2781980 and 2781984.
 
 If a call comes in to 2781980, and is active, and another call comes 
 in to 2781980, the second call will be busy.
 
 A call to 278-1984 will proceed while the 1980 is busy.
 
 The telco tells me though that the bri should be capable of hunting on

 it's own.
 
 I did this in the past with modem banks, but they were on top of 
 centrex.
 
 What I would like to do is put an 800 number to point to the 278-1980,

 and for the most part not use the 278-1984 except for maybe a disa.
 
 The eiconctrl monitor app is aware that the line is busy, and I do not

 believe it is notifying asterisk of the issue.
 
 I am trying to move some lines to bri since my audio quality on pots 
 has been horrible.  The isdn is great, especially since you told me of

 the ulaw modification I needed to make...  I got lucky with this one, 
 since they really could not install it without doing special 
 construction, which I managed to avoid paying the big bucks for 
 because the csr was nice about the 3 month delay.  I set it up through

 a panasonic dbs so the secretary can just hit a button, and I get 
 immediate rings on 4 sip phones and my cell.  I would love a PRI, but 
 only need 4 channels max which is why I went with the bri.
 
 Compared to pots, the isdn is way better.  I also find it much more 
 stable than IP, to the point where it is worth the 1c/minute to use.
 
 Thanks for the help.
 
 Greg
 
 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
 Sent: Tuesday, August 16, 2005 2:06 AM
 To: Gregory Wiktor - ADCom Corp.
 Subject: RE: [Asterisk-Users] BRI Hunting, using both channels on one 
 msn
 
 On ISDN, the second channel is automatically used if the first channel

 is busy.
 Normaly you never get a busy signal, just because ONE channel is busy.
 Only if there is no application/phone available for that MSN, then you

 get busy.
 Or maybe I just don't understand what you are doing...
 
 Armin
 
 On Mon, 15 Aug 2005 [EMAIL PROTECTED] wrote:
  Well,
  I want to direct a toll free to my first msn.  The problem is, if 
  the line is busy a busy signal is returned.  I want the line to hunt

  to the next channel, so it can be answered on the first msn.
  
  Regards,
  Greg
  
  -Original Message-
  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
  Sent: Monday, August 15, 2005 3:53 PM
  To: Gregory Wiktor - ADCom Corp.
  Cc: asterisk-users@lists.digium.com
  Subject: Re: [Asterisk-Users] BRI Hunting, using both channels on 
  one msn
  
  On Mon, 15 Aug 2005 [EMAIL PROTECTED] wrote:
   Hello All,
   Has anyone configured bri to answer for only one msn?  In essence,

   when the primary is busy I want to have channel 2 ring.
   
   I am using an eicon diva server bri
   
   I know I saw it in the windows interface, but don't see it in the 
   linux setup.
  
  This is normal behaviour. What exactly is your problem?
  
  Armin
  
 
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[Asterisk-Users] Is background() fax detect broken?

[gw]

Hello All,
I have noticed I cannot do a background() playback fax detect on the
latest cvs.  Has anyone experienced this?

Regards,
Greg
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RE: [Asterisk-Users] Asterisk ISDN: Problem Setting CallerID as DIDExtension Numbers.

[gw]

this happened to me on a cvs update, rebuilt a clean chan 
capi cm and all is well.

Greg


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Voicomm 
UserSent: Monday, September 19, 2005 3:29 AMTo: Armin 
SchindlerCc: asterisk-users@lists.digium.comSubject: Re: 
[Asterisk-Users] Asterisk ISDN: Problem Setting CallerID as DIDExtension 
Numbers.
Thanks very much Armin.After migrating to chan_capi-cm, the 
issue now is, everytime a dial statement is executed, it fails and restarts 
asterisk! The restart I believe is due to safe_asterisk script. So, in my 
opinion chan_capi-cm terminates asterisk process abruptly. When I replace 
the driver, everything comes back fine. I have updated the dial syntax to 
suit the new driver. My Dial command now is Dial(CAPI/g1/dialled #,30), 
I have even tried Dial(CAPI/contr1/dialled 
#,30)Unfortunately there arent many messges displayed on CLI, since 
asterisk getsa SIGSEGV.Have people in this list had any successfull 
implementation with chan_capi-cm driver and Eicon Hardware?Regards 
-r
On 8/28/05, Armin 
Schindler [EMAIL PROTECTED] 
wrote:
On 
  Sun, 28 Aug 2005, Voicomm User wrote: Hello Group, 
  Current Setup: - Eicon Quad BRI ISDN Card. - 4 ISDN BRI 
  (Telco: Telstra) Onramp2 services. - Mode: Point2Point. - 100 
  Indial Number ranges. Full National Number (9 digit format): BAAXX 
   where: B (Area code): 2/3/7/8 A (Normal 
  Numbers) X (99 Indial extensions) eg: BAA00 
  BAA20 etc Requirement: - To be able send Indial numbers as 
  Caller ID when dialing out.  Configration: 
  capi.conf - [general] 
  nationalprefix=0 internationalprefix=00 rxgain=0.8 
  txgain=0.8 [interfaces] mode=immediate  
  isdnmode=ptp msn=BAA incomingmsn=* 
  controller=1 softdtmf=1 accountcode= 
  context=main-menu usecallerid=yes callerid=asreceived 
  ;echosquelch=1  ;echocancel=yes ;echotail=64 
  callgroup=1 pickupgroup=1 devices=2 
  mode=immediate isdnmode=ptp msn=BAA 
  incomingmsn=* controller=2 softdtmf=1 
  accountcode= context=main-menu usecallerid=yes 
  callerid=asreceived callgroup=1 pickupgroup=1 
  devices=2 mode=immediate isdnmode=ptp  
  msn=BAA incomingmsn=* controller=3 
  softdtmf=1 accountcode= context=main-menu 
  usecallerid=yes callerid=asreceived callgroup=1 
  pickupgroup=1  devices=2 
  mode=immediate isdnmode=ptp msn=BAA 
  incomingmsn=* controller=4 softdtmf=1 
  accountcode= context=main-menu usecallerid=yes  
  callerid=asreceived callgroup=1 pickupgroup=1 
  devices=2 extensions.conf 
  [mob-service] ; Calleridnum is of the format 0BAAXX and TRUNKMSD = 
  1, TRUNKCAPI = CAPI  exten = 
  ${PAT-MOB},1,Dial(${TRUNKCAPI}/${CALLERIDNUM:1}:${EXTEN:${TRUNKMSD}},,t) 
  Problem: When dialling out the number *always* defaults to the default 
  service number.  I have contacted the telco and they have 
  confirmed they expect the caller id in 9 digit format. I tried 
  modifying msn value in capi.conf to include more comma 
  separated Full National Numbers of users internally. Eg.  
  msn=BAA00,BAA06,BAA07,BAA08,BAA09,BAA10,BAA11,BAA12,BAA13,BAA14,BAA15,BAA16,BAA17,BAA20,BAA21,BAA22 
  This works fine upto BAA17, but for numbers from extentions 20 onwards I 
   get a 'msn not found! check your config 
  error'. Can anyone please shed somelight on whether this is 
  really possible (to be able to send DID numbers as caller ID 
  when dialling out)? I have read some posts  indicating more than 5 
  msns is not possible, but in my case I have definetely got it 
  working with more than 5 msns atleast. No source clearly indicated if 
  this is possible, and if yes, how. Use chan_capi-cm from sourceforge.net, adapt your capi.conf and 
  thedialstring to new structure (see README of chan_capi-cm) and set your 
  DIDwith e.g.SetCallerId(15) 
Armin
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[Asterisk-Users] Monitor and sox mix quality

[gw]

Hello All,
I am using monitor with soxmix, however the quality seems somewhat low
after sox converts to mp3.

Does anyone know a way to get a higher quality file?  Some of my lines
are coming in on isdn.

Regards,
Greg
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[Asterisk-Users] capiFax causes segfault on asterisk

[gw]

Hello All,
I tried to implement capiFax to receive on an eicon diva server, and if
I call the msn, and hang up, the capifax starts but segfaults asterisk.

Also the system lags for about 5 seconds.

Anyone know how I can trace down the issue?

I am running head with chan_capi_cm

Regards,
Greg
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RE: [Asterisk-Users] capiFax causes segfault on asterisk

[gw]

Sorry I got it, needed to recompile a clean chan_capi-cm.

Regards,
Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gregory
Wiktor - ADCom Corp.
Sent: Saturday, September 17, 2005 3:18 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] capiFax causes segfault on asterisk

Hello All,
I tried to implement capiFax to receive on an eicon diva server, and if
I call the msn, and hang up, the capifax starts but segfaults asterisk.

Also the system lags for about 5 seconds.

Anyone know how I can trace down the issue?

I am running head with chan_capi_cm

Regards,
Greg
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RE: [Asterisk-Users] How to locate Toll Free Ownership

[gw]

I h ad a similar problem.  I have a number I want, which is unused.

Call verizon, ask them who the ld owning company is.  Mine ended up as
mci, then call them and ask, worth a shot...

Greg 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of calvis
Sent: Friday, September 02, 2005 1:41 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] How to locate Toll Free Ownership


How do you find out what company owns a certain 877 number?  Currently
it is disconnected and I have a friend that wants acquire it and port it
to my system.  I have Googled and have found people that do this for a
living, but surely there must be an easy way to find out without paying
a couple hundred dollars.

Sorry for this not really being Asterisk related, but there might be
some telecom people on the list that might point me in the right
direction.

Thanks,

-Calvis


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[Asterisk-Users] T1 DSU's/Split for voice

[gw]

Hello All,
I was wondering if I could do the following on asterisk...

Get a T1 between 2 locations, and split it into a data channel of like
1024, and use the rest for voice channels.

Has anyone done this and had it working well?  Or would I need to get a
csu that allows a split into two interfaces?

Regards,
Greg
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RE: [Asterisk-Users] Optimum online-upload throttling confirmed.

[gw]

Been there, done that...

I was talking to a high level tech for an hour...

Basically, they calculate the need for throttle based on the length of
time a modem is busy, not the amount of data that is transferred.

So for example, asterisk not involved, If I view an axis camera feed
remotely, after about 2 minutes the entire network lags.  Even though
it's only going 10-20k/second, it's the constant traffic that does it.

It's a cable thing, probably since they have so many modems up on their
nodes now... A year ago the node was nice and empty...

Regards,
Greg


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brendon
Baumgartner
Sent: Friday, August 19, 2005 12:45 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Optimum online-upload throttling
confirmed.

 From: [EMAIL PROTECTED] [mailto:asterisk-users- 
 [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
 Sent: Thursday, August 18, 2005 6:08 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Optimum online-upload throttling confirmed.
 
 Hello All,
 I was recently fighting with an optimum online connection in NY.
 
 I finally got in touch with someone that confirmed they are throttling

 my upload connection.
 
 I just wanted to make everyone aware of it, so if you have problems if

 your ping times jump erratically, this could be the cause.
 
 Their suggestions were, although you can upload a lot, do not do it 
 constantly.  They do not want any constant outgoing connections.
 
 Even on business class, they do throttle.  All business class 
 primarily does is allow port 25 to pass.
 
 Now I am going to look and see if I can get a decent upload speed dsl 
 or something to correct this problem.


You might try traffic shaping before going to your ISP. Being that ping
is erratic though, is evidence that it may not help.

I believe LARTC has some information for you there.

-Brendon

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[Asterisk-Users] Optimum online-upload throttling confirmed.

[gw]

Hello All,
I was recently fighting with an optimum online connection in NY.

I finally got in touch with someone that confirmed they are throttling
my upload connection.

I just wanted to make everyone aware of it, so if you have problems if
your ping times jump erratically, this could be the cause.

Their suggestions were, although you can upload a lot, do not do it
constantly.  They do not want any constant outgoing connections.

Even on business class, they do throttle.  All business class primarily
does is allow port 25 to pass.

Now I am going to look and see if I can get a decent upload speed dsl or
something to correct this problem.

Regards,
Greg
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