Re: [Asterisk-Users] SIP Registration --Giving up forever after very short network outage.

[Olle E. Johansson]

Steve Gladden wrote:
You also want to look at the "registertimeout" and "registerattempts"
> 
> 
> Yes!!!, thank you VERY much this is what I needed.
> Where are these options documented at?
> I'm guessing the source code?
> Or is there a better place to find this stuff?
> 
> A search on the wiki for "registertimeout" or "registerattempts"
> reveals absolutely nothing.
> 
> I had been searching ealier for things like SIP register timeout
> and "Giving up forever" all to no avail.
> 
You should always check configs/sip.conf.sample in your source code
directory. We update docs/ and configs/ very often.

We recently updated the behaviour on authentication for INVITEs as well
in CVS head, the base for 1.2. We will now give up if we can't
authenticate, so the call goes back to the dialplan with CONGESTION
instead of trying forever and ever.

/Olle

---
Astricon 2005 - wanna speak? Check http://www.astricon.net/2005/speakers
Looking for call center, business and service providers business cases!
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Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

[Lars Dybdahl]

You did not specify anything about your network. If your network has a
big latency, echo cancellers can get into trouble. For instance, I
have echo problems just using wireless POTS phones on my sipura 2100
sip adapter/router on an otherwise unused 8Mbps ADSL internet
connection at home.

Lars Dybdahl.

On 8/24/05, canuck15 <[EMAIL PROTECTED]> wrote:
> My problem is that I cannot eliminate echo no matter what I try.
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Re: [Asterisk-Users] ASTCC and cdrs

[Darren Wiebe]

When did you install it?  Try running the "update database" function 
from the configure menu.


Darren Wiebe
[EMAIL PROTECTED]

Il Neofita wrote:


My installation of ASTCC does not update the cdrs tables .
It is a problem of ASTCC or it is a configuration problem?
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Re: [Asterisk-Users] Error when answering CAPI

[Armin Schindler]

On Wed, 24 Aug 2005, Humberto Aicardi wrote:
> Hi,
> 
>I've a Fritz card which was working fine, recently I changed hardware and
> my nightmare started. Now when I call someone through the chan_capi (0.3.5 or
> 0.4.0) it works fine but when I receive calls I always get hungup. Can someone
> please give some help? Here are the logs:
> 
> *CLI>
>  -- CONNECT_IND ID=001 #0x LEN=0049
> Controller/PLCI/NCCI= 0x101
...

The logs are incomplete to say something. I don't remember if a higher 
verbose level will show the necessary parts.
I suggest you use chan_capi-cm from sourceforge.
With verbose level 5 and 'capi debug' it shows everything you need.

Armin

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Re: [Asterisk-Users] chan_capi, modprobe: Can't locate module capifs, ERROR: fopen(/etc/capi.conf, r)

[Armin Schindler]

On Thu, 25 Aug 2005, Goran Dj. wrote:
> > But, now I cannot start chan_capi.so:
> > WARNING[802]: chan_capi.c:3106 load_module: CAPI not installed, CAPI
> > disabled!
> > 
> > from tty:
> > capiinit
> > ERROR: cannot open /dev/capi20 nor /dev/isdn/capi20 - No such file or
> > directory (2)
> > 
> > capiinfo
> > capi not installed - No such file or directory (2)
> > 
> > capiinit show
> > ERROR: fopen(/etc/capi.conf,r) failed - No such file or directory (2)
> > ERROR: no cards configured in /etc/capi.conf
> 
> 
> I resolved missing /dev/capi20 with shell script "makedev-capi.sh"
> but, now, when starting capiinit:
> 
> modprobe: Can't locate module capifs
> modprobe: Can't locate module capifs
> WARNING: filesystem capifs not available
> ERROR: fopen(/etc/capi.conf,r) failed - No such file or directory (2)

capifs is not necessary. You only need this when you want to use e.g.
pppd with the pppdcapiplugin.
This is an option in the kernel config.

Anyway, just installing the capi drivers isn't enough, because your isdn 
card must provide a capi interface (which is then connected to kernelcapi).
As far as I know, your card is supported by the HiSax driver, but this 
driver does not provide capi interface, only the 'old' isdn4linux interfaces
like ttyI are supported.
The new mISDN does provide CAPI, but I don't know the status of the support 
for your card.

/etc/capi.conf and capiinit is used by AVM cards only. It's useless for you.

Armin

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[Asterisk-Users] Re: Motherboards and IRQs

[jennyw]

Colin Anderson wrote:

Get ye to a used computer store and get one of these:

http://cgi.ebay.com/Compaq-Deskpro-EN-Desktop-550MHZ-128MB-6-4GB_W0QQitemZ52
33796857QQcategoryZ51114QQrdZ1QQcmdZViewItem

[...]

cat5 and setting up a telemarketer honeypot. Hey, for $50, what do you have
to loose? 


Thanks! I might try that, although I'm not sure what my chances are of 
finding that particular model at a used computer store.


Trouble is, I'm leaving for an out of town trip early next week and want 
to get things settled before then. So I was hoping to drop by the 
computer store tomorrow and pick up what I need, so was hoping to find 
the name of a board that could assign IRQs that would be popular enough 
that it'd likely be in stock ...


Jen


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Re: [Asterisk-Users] asterisk in Taiwan

[Julian Chen]

On 8/24/05, Lance Grover <[EMAIL PROTECTED]> wrote:
> Is there someone out there that is using Chunghwa Telecom in Taiwan
> with asterisk?  I have one there connecting using a digium 4 port fxo
> card.  We can get and make calls. the problem is that about 40% of the
> time when we make a call - even if it is one that we have made
> recently - we get back "That number can not be compleeted as dialed"
> or something like that in Chinese.
> 
> Any help would be nice - and my zaptel.conf does have language=tw and
> defaultlanguage=tw.
> 

I've installed one [EMAIL PROTECTED] 1.1 in June with TDM11B (rev.H) and
worked fine.  I just tested one week, about 20 calls/day.

in my zaptel.conf, I used: loadzone=tw and defaultzone=tw
in zapata.conf, usecallerid=no

No other changes from default installation.

Regards,

Julian
 
-- 
Use the force, read the source!
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RE: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

[canuck15]

 
Colin,

Your suggestions about identical hardware and BIOS revisions sound like good
advice.  If I ever get past this it is something I will definitely be
careful about.

Well I am starting to think that maybe it is my PC. I have a newer, faster
AMD system with PCI2.2 so I might give that one a shot.

You are using FXS modules with analog phones.  I am using SIP phones.  It is
my understanding that there is a LOT more delay when using SIP phones and of
course a lot of other things going on.

zttest yields 100% at least half the time and  99.987793% the rest of the
time.

Someone else suggested enabling MMX and CONFIG CALC XLAW in zconfig.h as
well as adding the following to the Zaptel Makefile:
KFLAGS+=-march=pentium3
CFLAGS+=-march=pentium3

I did all this, recompiled zaptel and rebooted but it didn't seem to make
any difference.

-Original Message-
From: Colin Anderson [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 24, 2005 8:18 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

dude it's gotta be something with your system. Im using same setup at home
with a TDM22 with no probs. [EMAIL PROTECTED] 1.5, Compaq Deskpro EN P3 500,
cordless phones.  You did a ton more than I did, I basically plugged
everything in and installed [EMAIL PROTECTED] first try. 

Hate to say it, but would it be possible to try a different system? Even an
old system as long as it is PCI 2.2.

The big clue here is that you say that echo is the same regardless of phone
or PSTN source. That pretty much narrows it down to the system. What's your
score on /usr/src/zaptel/zttest?

This is the burn of Asterisk. It's extremely difficult to make random
hardware work good with it. Because it is designed for commodity hardware,
stability is a moving target. Even bios revs can make a difference. In the
deployment I have done for my work I have a master Asterisk server and 30
IAX servers in remote locations. All 30 boxen are exactly the same, down to
the BIOS rev, Linux version, network cards, everything. This is the way to
do it, to narrow down infinite variables to a controllable few.  You need
patience or luck but preferably both. You have an advantage, though. From
your post it looks like you've done your homework, you aren't a dumbass, and
you know how to arrive at a conclusion through process of elimination.
You're on the right track. Don't give up now, 'cause you are close.

> --
> From: canuck15
> Reply To: Asterisk Users Mailing List - Non-Commercial Discussion
> Sent: Wednesday, August 24, 2005 5:02 PM
> To:   'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject:  RE: [Asterisk-Users] Will Echo problems EVER be solved, I'm
> scared
> 
> <>
> Wiley,
>  
> The very first thing I checked was for IRQ problems.  I apoligize for 
> forgetting to mention that.
>  
> The only thing I found in the Google search you suggested is a thread 
> from January 2004 suggesting that businesses should ONLY use a T1 card 
> with a channel bank instead of X100P cards.  I understand the "don't use
X100P"
> part but I just assumed the TDM400P with echo cancellation is working 
> fine now.
>  
> I am using a WRT54G (switch).  I am NOT connecting via wireless.  I 
> have no traffic to QoS.  Just VoIP and a WRT54G switch is quite 
> capable of that as far as I know.  No hubs!
>  
> I pretty much just use AAH now for it's ease of install but I have 
> rolled my own in the past.  The echo has persisted through AAH v1.3, 
> 1.4, 1.5.  I have recently (yesterday) installed the latest Asterisk 
> and Zaptel CVS head development tree just to see if it made a difference
and it DID NOT!
>  
>  
>  
>  
> 
>   _
> 
> From: Wiley Siler [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, August 24, 2005 2:00 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Will Echo problems EVER be solved, I'm 
> scared
> 
> 
> Just because you cannot get it to work does not mean that IT does not 
> work.
>  
> Just using the right motherboard is not enough.  Did you check for IRQ 
> problems?  You don't mention whether you have checked for this.
> Look for a thread called "Asterisk-Users Small office setupusing 
> analog lines w Asterisk" in the archive via Google.
> use site:lists.digium.com
> Try all the things listed in that thread.
>  
> Do you have a network that is capable of VoIP?  Are you using hubs 
> when you should be using switches?
> There is a major difference and hubs WILL NOT work reliably with VoIP. 
> Are you using QoS on your switches if you have lots of network traffic?
>  
> If you are using your own Distro and installing from scratch, try to 
> use Asterisk at Home just to see if you still have the same problem.
>  
> I am putting my money on an IRQ issue myself.
>  
> W
>  
>  
>  
>  
>  
> 
>   _
> 
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of 

Re: [Asterisk-Users] Fwd: asterisk in Taiwan

[Keith Caldwell]

I've never set up asterisk in Tiawan but I had a few issues like that  
here in the U.S. I solved it by putting a pause in the dail command  
so that asterisk could fully open the channel before it started to dial.


exten => _91800NXX,1,ChanIsAvail(Zap/4)
exten => _91800NXX,2,Dial(zap/4/ww${EXTEN:1}) */note the w
exten => _91800NXX,102,congestion


Good luck

Keith


On Aug 24, 2005, at 9:55 PM, Lance Grover wrote:


No responce?

How about if anyone has even setup a asterisk server in Taiwan
connecting to an analog phone line


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[Asterisk-Users] Asterisk hint thing.... what do you do with it?

[Steve Gladden]

I'm having difficulty understanding this 'hint' feature of asterisk.

My limited understanding is that it is somehow needed for 'informing'
some kinds of phones that can do shared line appearance to show the
state of the channel/user...
Is this true?

the wiki has this:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20presence

I'm still having much difficulty getting a basic understanding of what this
is what it does how it works and how to debug/test it to see that it is
working...

:-)

What I'd like to accomplish is to get some snom phones and have shared call
appearance lights go on & off when certain extensions are in use etc.

Is this hint thing what is needed?  is there a way to test it without
actually using a phone that has the shared call appearances?

I tried something like this:
exten => 4300,hint,SIP/4300
exten => 4300,2,Dial(SIP/4300,33,tT)
exten => 4300,3,Voicemail(u4300)
exten => 4300,4,hangup



set verbose to 4 and sip debug on...

I don't see anything different when extensions 4300 is picked up
dialing or connecting a call

however doing a show hints at the CLI shows a state chage from 0 to 1 when
the extension is in use.

Any pointers on how to use/test the hint feature would be greatly
appreciated!!

Take care!

Steve






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Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

[Matt Fredrickson]

Ok, fxotune is a work in progress so to speak.  I fixed something in it about
a week ago that may help it adjust to the line better (whereas before I'm not
sure that it was at all).  Try the latest CVS-HEAD version of fxotune as your
first step.  (oh, after you use fxotune you should turn off your gain settings
in zapata.conf).

Second step is to try the new echo canceller that was added to CVS-HEAD.  Look
in zconfig.h and try the KB1 echo canceller.  I have received many good reports
that it has cured practically all echo on all of the systems that I have heard
feedback from.

If all of this doesn't work, you probably have a serious hardware line issue
that you should resolve with your telco.

---
Matthew Fredrickson
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Re: [Asterisk-Users] SIP Registration --Giving up forever after very short network outage.

[Steve Gladden]

>>>You also want to look at the "registertimeout" and "registerattempts"

Yes!!!, thank you VERY much this is what I needed.
Where are these options documented at?
I'm guessing the source code?
Or is there a better place to find this stuff?

A search on the wiki for "registertimeout" or "registerattempts"
reveals absolutely nothing.

I had been searching ealier for things like SIP register timeout
and "Giving up forever" all to no avail.

Steve












> On 8/24/05, Eric Wieling aka ManxPower <[EMAIL PROTECTED]> wrote:
>> Try using IP addresses instead of hostnames in sip.conf.  Asterisk's DNS
>> support is supposed to be improved in CVS-HEAD, but you should still try
>> it.
>>
>> However, using an IP address instread of a hostname in your host= line
>> could have issues with some ways a provider might do failover and load
>> balancing.
>
> You also want to look at the "registertimeout" and "registerattempts"
> options for your sip.conf. I had lots of problem staying registered
> with various providers, so now I'm running with "registerattempts=0",
> IOW try forever to (re-)register. In conjunction with the
> registertimeout you have some control over how often you retry. (IIRC,
> both options are CVS-HEAD only, not available in stable. But so is the
> "Giving up forever" error. At least I think that's the case.)
>
> --
> "I am Dyslexic of Borg. Fusistance is retile. Your ass will be lamitated!"
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[Asterisk-Users] Fwd: asterisk in Taiwan

[Lance Grover]

-- Forwarded message --
>From: Lance Grover <[EMAIL PROTECTED]>
>Date: Aug 24, 2005 8:54 AM
>Subject: asterisk in Taiwan
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>
>
>
>Is there someone out there that is using Chunghwa Telecom in Taiwan
>with asterisk?  I have one there connecting using a digium 4 port fxo
>card.  We can get and make calls. the problem is that about 40% of the
>time when we make a call - even if it is one that we have made
>recently - we get back "That number can not be compleeted as dialed"
>or something like that in Chinese.
>
>Any help would be nice - and my zaptel.conf does have language=tw and
>defaultlanguage=tw.
>

No responce?

How about if anyone has even setup a asterisk server in Taiwan
connecting to an analog phone line?


--
Thanks,

Lance Grover


-- 
Thanks,

Lance Grover
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Re: [Asterisk-Users] HOW TO SEND A MESSAGE TO A CHANNEL THAT IS RECEIVING A CALL????

[Matt Riddell]

j_amorim wrote:
> Dear guys, 
> 
> Can you give a tip to solve this problem. 

Option 1: Answer a second call with predefined callerID, receive text on it -
requires that you have control of the source code.

Option 2: Hangup on the call

:)

I really think that IAX devices should be able to send/receive text messages
without having to actually be in a call.

-- 
Cheers,

Matt Riddell
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Re: [Asterisk-Users] AGI + Ruby

[Innocent Evil]

Thanks guys, for your great humors, joke and fun writes ..

Well this link helped to start writing AGI in Ruby (An object oriented
scripting language, for the people who dont knows Ruby is only a expensive
stone !!)

http://home.cogeco.ca/~camstuff/agi.html

Thanks again,




> -Original Message-
> From: [EMAIL PROTECTED]
> Sent: Wed, 24 Aug 2005 17:48:05 -0400
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] AGI + Ruby
>
> YEah but the problem with pearls are that they come in different colours
> and are often of varying quality.
>
> Black one (which are actually green) are the best.
>
>
>
> Huddleston, Robert wrote:
> > Actually Perl is even better
> >
> >
> >
> >
> >>-Original Message-
> >>From: [EMAIL PROTECTED]
> >>[mailto:[EMAIL PROTECTED] On Behalf Of
> >>Mark Phillips
> >>Sent: Wednesday, August 24, 2005 4:17 PM
> >>To: Asterisk Users Mailing List - Non-Commercial Discussion
> >>Subject: Re: [Asterisk-Users] AGI + Ruby
> >>
> >>Y'see it? There it goes! Right over his head.
> >>
> >>
> >>Huddleston, Robert wrote:
> >>
> >>>U joke - duh!
> >>>
> >>>
> >>>
> >>>
> >>>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf
> >>
> >>Of Innocent
> >>
> Evil
> Sent: Wednesday, August 24, 2005 3:53 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] AGI + Ruby
> 
> What IDE are you talking about?
> Any URL would be helpful.
> 
> Thanks,
> 
> 
> 
> 
> 
> >-Original Message-
> >From: [EMAIL PROTECTED]
> >Sent: Wed, 24 Aug 2005 15:16:18 -0400
> >To: asterisk-users@lists.digium.com
> >Subject: Re: [Asterisk-Users] AGI + Ruby
> >
> >I think you might find amethyst much simpler and possibly
> 
> cheaper too.
> 
> 
> >I believe the current IDE is 12.4K
> >
> >
> >
> >Innocent Evil wrote:
> >
> >
> >>I would like to write AGI script in Ruby Would anybody
> 
> please show
> 
> 
> >>me right direction..
> >>
> >>
> >>Thanks___
> >>Asterisk-Users mailing list
> >>Asterisk-Users@lists.digium.com
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>To UNSUBSCRIBE or update options visit:
> >>  http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >
> >--
> >
> >Mark, G7LTT/KC2ENI
> >Randolph, NJ
> >http://www.g7ltt.com
> >___
> >Asterisk-Users mailing list
> >Asterisk-Users@lists.digium.com
> >http://lists.digium.com/mailman/listinfo/asterisk-users
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> >
> >
> 
> http://lists.digium.com/mailman/listinfo/asterisk-users_
> >
> > __
> >
> >
> 
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> >>>
> >>>___
> >>>Asterisk-Users mailing list
> >>>Asterisk-Users@lists.digium.com
> >>>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>>To UNSUBSCRIBE or update options visit:
> >>>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >>>
> >>
> >>--
> >>
> >>Mark, G7LTT/KC2ENI
> >>Randolph, NJ
> >>http://www.g7ltt.com
> >>___
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> >>Asterisk-Users@lists.digium.com
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> --
>
> Mark, G7LTT/KC2ENI
> Randolph, NJ
> http://www.g7ltt.com
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RE: [Asterisk-Users] Motherboards and IRQs

[Colin Anderson]

Get ye to a used computer store and get one of these:

http://cgi.ebay.com/Compaq-Deskpro-EN-Desktop-550MHZ-128MB-6-4GB_W0QQitemZ52
33796857QQcategoryZ51114QQrdZ1QQcmdZViewItem

Your original post said you were trying to get a small office going. This
would be suitable for a dozen or so users as long as you weren't
transcoding. 

When you boot it, hit F10 and go into the BIOS. In Advanced Setup > PCI you
can set any PCI card to any IRQ you want. A TDM card shows up as "Unknown
Intel network controller". I set my TDM22 to IRQ 11 and disabled the onboard
audio, parallell, etc to give plenty of free IRQ's. I set the Intel NIC to
IRQ 7, for example, the same IRQ as the parallel port. 

On mine, zttest yields 100% about 70% of the time, no echo, and with 3
extensions going, top says CPU at ~10%. It took me a couple hours total to
get it going first time with [EMAIL PROTECTED] and that includes terminating the
cat5 and setting up a telemarketer honeypot. Hey, for $50, what do you have
to loose? 

> --
> From: jennyw
> Reply To: Asterisk Users Mailing List - Non-Commercial Discussion
> Sent: Wednesday, August 24, 2005 5:39 PM
> To:   Asterisk Users Mailing List - Non-Commercial Discussion
> Subject:  [Asterisk-Users] Motherboards and IRQs
> 
> Someone mentioned earlier (I can't find the message now) that they had a 
> motherboard that allowed you to change IRQ assignments in BIOS. Does 
> anyone happen to know how to identify motherboards that can do this? I'm 
> going to put together a new machine now and I'm having trouble picking a 
> motherboard for it (ordering from Dell or other online vendor is not an 
> option, since I need this in the next couple days).
> 
> This is to provide another avenue for avoiding IRQ conflicts with the 
> Digium TDM400Ps.
> 
> Thanks!
> 
> Jen
> 
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[Asterisk-Users] SIP trunk rollover problem

[Chris Miller]

Hello,
	I've got an Asterisk system with 3 SIP trunks configured. Each SIP 
trunk is actually a 4 port Mediatrix PSTN gateway. The current outbound 
call routing (via AMP 1.10.007a) uses the 3 trunks in descending order, 
all set with max channels to 4. Unfortunately, when the first trunk 
reports a "480 Service Unavailable" (all ports in use), Asterisk reports 
congestion without rolling to the next available trunk.


	I've looked at the AMP dialplan on this system, as well as a more 
recent system (1.10.009beta1), and although the dialplan has been 
improved, it still doesn't seem to account for this condition. As you 
can see below at step 7, if the max channels have been used on the 
current trunk, the call fails.


	What is the correct way to do fail over between trunks, and in an AMP 
friendly way that won't get clobbered during the next config change?


Regards,
Chris

[macro-dialout-trunk]
exten => s,1,GotoIf($[foo${ARG3} = foo]?3:2))   ; arg3 is pattern password
exten => s,2,Authenticate(${ARG3})
exten => s,3,Macro(user-callerid)
exten => s,4,Macro(record-enable,${CALLERIDNUM},OUT)
exten => s,5,Macro(outbound-callerid,${ARG1})
exten => s,6,SetGroup(OUT_${ARG1})
exten => s,7,CheckGroup(${OUTMAXCHANS_${ARG1}})
; if we've used up the max channels, continue at 110 (n+101)
exten => s,8,SetVar(DIAL_NUMBER=${ARG2})
exten => s,9,SetVar(DIAL_TRUNK=${ARG1})
exten => s,10,AGI(fixlocalprefix) ; this sets DIAL_NUMBER to the proper 
dial string for this trunk
exten => s,11,SetVar(OUTNUM=${OUTPREFIX_${ARG1}}${DIAL_NUMBER})  ; 
OUTNUM is the final dial number
exten => s,12,Cut(custom=OUT_${ARG1},:,1)  ; Custom trunks are prefixed 
with "AMP:"

exten => s,13,GotoIf($[${custom} = AMP]?16)
exten => s,14,Dial(${OUT_${ARG1}}/${OUTNUM})  ; Regular Trunk Dial
exten => s,15,Goto(s-${DIALSTATUS},1)

exten => s,110,Noop(max channels used up)
exten => s-BUSY,1,NoOp(Trunk is reporting BUSY)
exten => s-BUSY,2,Busy()
exten => s-BUSY,3,Wait(60)
exten => s-BUSY,4,NoOp()

exten => _s-.,1,NoOp(Dial failed due to ${DIALSTATUS})
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Re: [Asterisk-Users] dingotel - connect Asterisk to 2-way radio?

[Mark Phillips]

This has been round a few times.

The bottom line is that it's not doable with this product. The nearest 
solution is either app_rpt which requires a bespoke board to run the 
radio using * as a repeater controller with voip links or to use a phone 
patch device plugged into an ATA. The later can handle the simplex 
nature of an FRS/GMRS type radio as it uses a crude noise gate to 
operate the TX. When there is noise (speech/DTMF/whatever) on the line 
side the TX activates otherwise it relays the radio side.


You should also be aware of the legality in your part of the world for 
this type of system. Many PTT's don't allow connection of radio devices 
to the PSTN or Internet either at all or without a waiver.


In the US it is legal on ham bands and GMRS (460MHz) provided the 
operator has "control" and not at all on MURS (156MHz) and FRS (460MHz 
interleaved with GMRS). In the UK its only legal on the ham bands and 
PMR446 as long as the operator is physically present at the RF/Internet 
location. YMMV


Mark, G7LTT/KC2ENI

Shawn Rutledge wrote:
So has anybody got one of these?  


http://www.amazon.com/exec/obidos/ASIN/B0007LQQUK/qid%3D1106972010/sr%3D11-1/ref%3Dsr%5F11%5F1/102-1529886-6420131

I'm thinking that it should be possible to connect it directly to an
Asterisk box and not use their software, as long as there was Linux
support for the USB dongle.  Maybe it just looks like a standard USB
audio device?  But there has to be an extra ring on the connector to
"key up" the radio so maybe to support that, the usual USB audio
driver would need some customization.  Just wondering if anybody has
tried it in Linux.

If you could do that, then Asterisk could listen for DTMF tones right?
 So you could use it from any 2-way radio which has a keypad, such as
a ham 2-meter rig.  (Or hack a keypad into your FRS or CB radio.)  Or,
Asterisk could be tied to a speech recognition system (like their
software has) but that is more complex.

Another aspect is the logic that is required to decide when to
transmit and when to listen.  You'd need to recognize the presence or
absence of human voice from both the VOIP end and the radio end, and
try to balance the two people's rights to monopolize the conversation.
 The limitation is that the radio is not full-duplex, like a repeater,
so when the person on the voip end is talking, and the radio whcih is
attached to the dingotel is transmitting that signal, it is not
listening, and nothing that the person on the remote radio end says is
going to get through.  It could be improved by actually using a
full-duplex radio though.
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--

Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
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RE: [Asterisk-Users] Will Echo problems EVER be solved, I'm scare d

[Colin Anderson]

dude it's gotta be something with your system. Im using same setup at home
with a TDM22 with no probs. [EMAIL PROTECTED] 1.5, Compaq Deskpro EN P3 500,
cordless phones.  You did a ton more than I did, I basically plugged
everything in and installed [EMAIL PROTECTED] first try. 

Hate to say it, but would it be possible to try a different system? Even an
old system as long as it is PCI 2.2.

The big clue here is that you say that echo is the same regardless of phone
or PSTN source. That pretty much narrows it down to the system. What's your
score on /usr/src/zaptel/zttest?

This is the burn of Asterisk. It's extremely difficult to make random
hardware work good with it. Because it is designed for commodity hardware,
stability is a moving target. Even bios revs can make a difference. In the
deployment I have done for my work I have a master Asterisk server and 30
IAX servers in remote locations. All 30 boxen are exactly the same, down to
the BIOS rev, Linux version, network cards, everything. This is the way to
do it, to narrow down infinite variables to a controllable few.  You need
patience or luck but preferably both. You have an advantage, though. From
your post it looks like you've done your homework, you aren't a dumbass, and
you know how to arrive at a conclusion through process of elimination.
You're on the right track. Don't give up now, 'cause you are close.

> --
> From: canuck15
> Reply To: Asterisk Users Mailing List - Non-Commercial Discussion
> Sent: Wednesday, August 24, 2005 5:02 PM
> To:   'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject:  RE: [Asterisk-Users] Will Echo problems EVER be solved, I'm
> scared
> 
> <> 
> Wiley,
>  
> The very first thing I checked was for IRQ problems.  I apoligize for
> forgetting to mention that.
>  
> The only thing I found in the Google search you suggested is a thread from
> January 2004 suggesting that businesses should ONLY use a T1 card with a
> channel bank instead of X100P cards.  I understand the "don't use X100P"
> part but I just assumed the TDM400P with echo cancellation is working fine
> now.
>  
> I am using a WRT54G (switch).  I am NOT connecting via wireless.  I have
> no traffic to QoS.  Just VoIP and a WRT54G switch is quite capable of that
> as far as I know.  No hubs!
>  
> I pretty much just use AAH now for it's ease of install but I have rolled
> my own in the past.  The echo has persisted through AAH v1.3, 1.4, 1.5.  I
> have recently (yesterday) installed the latest Asterisk and Zaptel CVS
> head development tree just to see if it made a difference and it DID NOT!
>  
>  
>  
>  
> 
>   _  
> 
> From: Wiley Siler [mailto:[EMAIL PROTECTED] 
> Sent: Wednesday, August 24, 2005 2:00 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Will Echo problems EVER be solved, I'm
> scared
> 
> 
> Just because you cannot get it to work does not mean that IT does not
> work.  
>  
> Just using the right motherboard is not enough.  Did you check for IRQ
> problems?  You don't mention whether you have checked for this.
> Look for a thread called "Asterisk-Users Small office setupusing analog
> lines w Asterisk" in the archive via Google.
> use site:lists.digium.com
> Try all the things listed in that thread.
>  
> Do you have a network that is capable of VoIP?  Are you using hubs when
> you should be using switches?
> There is a major difference and hubs WILL NOT work reliably with VoIP. 
> Are you using QoS on your switches if you have lots of network traffic?
>  
> If you are using your own Distro and installing from scratch, try to use
> Asterisk at Home just to see if you still have the same problem.
>  
> I am putting my money on an IRQ issue myself.
>  
> W
>  
>  
>  
>  
>  
> 
>   _  
> 
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of canuck15
> Sent: Wednesday, August 24, 2005 1:38 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared
> 
> 
>  
> I came into this with my eyes wide open.  I have read ABSOLUTELY
> EVERYTHING there is to be found on the net about avoiding echo problems
> BEFORE I even attempted to create a production system.  Since lots of
> people are apparently using this in production environments now I just
> assumed that echo IS avoidable.  
>  
> As others have recommended, I created a test system with the proposed
> production parts.  I bought a couple different SIP phones to try and a
> Digium TDM01B card.  I am using an older PIII 1Ghz system with 815chipset
> (PCI Rev2.2) with 256MB for my test system.  The only thing that will be
> different on a production system is that I will be using a newer chipset
> PC with faster processor and 512MB.  Probably Intel 7505, 7210, or 7211
> chipsets which seem to be the most compatible with Asterisk.  
>  
> My problem is that I cannot eliminate echo no matter what I try.  I
> seriously doubt that 

Re: [Asterisk-Users] Motherboards and IRQs

[Michael Welter]

Paul wrote:

jennyw wrote:

Someone mentioned earlier (I can't find the message now) that they had 
a motherboard that allowed you to change IRQ assignments in BIOS. Does 
anyone happen to know how to identify motherboards that can do this? 
I'm going to put together a new machine now and I'm having trouble 
picking a motherboard for it (ordering from Dell or other online 
vendor is not an option, since I need this in the next couple days).


This is to provide another avenue for avoiding IRQ conflicts with the 
Digium TDM400Ps.


Thanks!

Jen



intel 875p chipset and ECC RAM works well.

I have built several servers with intel and abit boards using this chipset

Always use a good UPS rather than a surge protector outlet strip. You 
can get an APC brand with USB signalling for as little as $50. It works 
with linux. That means the linux server will see when the battery is 
low, prepare itself for shutdown and then tell the APC to kill its 
output. When power returns the APC will turn outpput back on and the 
linux server boots gracefully.


When buying CPU I usually go with the "retail boxed" instead of OEM. It 
doesn't cost much more. It includes the cooler along with a good 
warranty. If it fails you get a new cooler along with the new chip. 
Obviously they can't try to get out of warranty replacement based on the 
cooler selection if it was their choice in the first place.


I stopped using athlons when I could no longer get boards with the AMD 
chipset(supports ECC). You can probably still get boards designed for 
dual athlon. I asked AMD why and they told me the 760 chipset for single 
athlon was no longer in production. It seems the average consumer 
doesn't appreciate higher quality features enough. It's too bad because 
I always liked AMD stuff. When I needed fast reliable chips for some 
specialized processors I designed it was AMD that gave me the first 
samples and they always gave the purchasing people good pricing for 
production needs. I'm only mentioning this because there has been some 
banter about AMD vs. Intel here. If the right motherboards were 
available for single socket 7 servers I would have stayed with them. 
Their service is excellent. They give me free overnight shipping and 2 
weeks to send in the defective part.



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Try the Tyan S2850--single Opteron.
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[Asterisk-Users] Distorted Sound from E1

[George Pajari]

We're having a problem with an E1 trunk in Mexico into an IVR server and 
would appreciate any suggestions.


Hardware: Digium TE110P jumpered for E1
zaptel.conf:
   span=1,1,0,ccs,hdb3
   # clear=1-30
   bchan=1-15
   bchan=17-31
   dchan=16
   loadzone = us
   defaultzone=us

Circuit status is fine: Status: Provisioned, Up, Active
Calls are accepted by Asterisk without any errors/alarms.
Audio from Asterisk (i.e. playback of pre-recorded sounds is fine).
Audio to Asterisk (i.e. sounds and DTMF from people calling Asterisk 
over the E1) sound as if played back on a tape recorder running at 
half-speed -- slow and down an octave.


What could be causing this?

--
George Pajari, netVOICE communications604 484 VOIP (484 8647 x102)
Open Source VoIP/Telephony Specialists  1 877 NET VOIP (638 8647 x102)
 www.netvoice.ca  www.ip-centrex.ca
 www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca

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Re: [Asterisk-Users] SIP Registration --Giving up forever after very short network outage.

[Kai-Uwe Jensen]

On 8/24/05, Eric Wieling aka ManxPower <[EMAIL PROTECTED]> wrote:
> Try using IP addresses instead of hostnames in sip.conf.  Asterisk's DNS
> support is supposed to be improved in CVS-HEAD, but you should still try it.
> 
> However, using an IP address instread of a hostname in your host= line
> could have issues with some ways a provider might do failover and load
> balancing.

You also want to look at the "registertimeout" and "registerattempts"
options for your sip.conf. I had lots of problem staying registered
with various providers, so now I'm running with "registerattempts=0",
IOW try forever to (re-)register. In conjunction with the
registertimeout you have some control over how often you retry. (IIRC,
both options are CVS-HEAD only, not available in stable. But so is the
"Giving up forever" error. At least I think that's the case.)

-- 
"I am Dyslexic of Borg. Fusistance is retile. Your ass will be lamitated!"
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Re: [Asterisk-Users] SIP Registration --Giving up forever after very short network outage.

[Eric Wieling aka ManxPower]

Steve Gladden wrote:

I'm looking for some help in how to keep asterisk from doing this.
If we loose Internet or routing to our upstream provider even for only a
few short minutes asterisk quickly gives up & never tries again.
I have to do a manual reload to get it to register with my
sip provider(s) again before incoming calls are accepted.


Try using IP addresses instead of hostnames in sip.conf.  Asterisk's DNS 
support is supposed to be improved in CVS-HEAD, but you should still try it.


However, using an IP address instread of a hostname in your host= line 
could have issues with some ways a provider might do failover and load 
balancing.

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[Asterisk-Users] Digium TDM400 in UK with BT Lines

[Graham Kiff]

Title: Message



Can anyone confirm 
they have a TDM400 connected for 1 or more BT lines running under Asterisk in 
the UK?
If so, what cable 
have you used to go from the BT socket to the back of the Digium card (in my 
case it has RJ11 sockets).
 
Cheers
Graham
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Re: [Asterisk-Users] Motherboards and IRQs

[Paul]

jennyw wrote:

Someone mentioned earlier (I can't find the message now) that they had 
a motherboard that allowed you to change IRQ assignments in BIOS. Does 
anyone happen to know how to identify motherboards that can do this? 
I'm going to put together a new machine now and I'm having trouble 
picking a motherboard for it (ordering from Dell or other online 
vendor is not an option, since I need this in the next couple days).


This is to provide another avenue for avoiding IRQ conflicts with the 
Digium TDM400Ps.


Thanks!

Jen


intel 875p chipset and ECC RAM works well.

I have built several servers with intel and abit boards using this chipset

Always use a good UPS rather than a surge protector outlet strip. You 
can get an APC brand with USB signalling for as little as $50. It works 
with linux. That means the linux server will see when the battery is 
low, prepare itself for shutdown and then tell the APC to kill its 
output. When power returns the APC will turn outpput back on and the 
linux server boots gracefully.


When buying CPU I usually go with the "retail boxed" instead of OEM. It 
doesn't cost much more. It includes the cooler along with a good 
warranty. If it fails you get a new cooler along with the new chip. 
Obviously they can't try to get out of warranty replacement based on the 
cooler selection if it was their choice in the first place.


I stopped using athlons when I could no longer get boards with the AMD 
chipset(supports ECC). You can probably still get boards designed for 
dual athlon. I asked AMD why and they told me the 760 chipset for single 
athlon was no longer in production. It seems the average consumer 
doesn't appreciate higher quality features enough. It's too bad because 
I always liked AMD stuff. When I needed fast reliable chips for some 
specialized processors I designed it was AMD that gave me the first 
samples and they always gave the purchasing people good pricing for 
production needs. I'm only mentioning this because there has been some 
banter about AMD vs. Intel here. If the right motherboards were 
available for single socket 7 servers I would have stayed with them. 
Their service is excellent. They give me free overnight shipping and 2 
weeks to send in the defective part.



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Re: [Asterisk-Users] Someone can explain alerts in zap channels?

[Andrew Kohlsmith]

On Wednesday 24 August 2005 19:43, Carlos Trallero wrote:
>  Can someone explain me the meaning of a red alarm in
> a zap channel. Is there anyway of removing it?

With the X101P (what I'm guessing you're using) a red alarm is generated when 
it senses no DC voltage from the telco.  This tells Asterisk that the port 
has no phone line connected to it.

-A.
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Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

[Andrew Kohlsmith]

First off, thank you *very* much for this unbelievably informative post!  I've 
got it saved away now along with Kris Boutilier's adjusting rxgain/txgain 
post.

On Wednesday 24 August 2005 17:14, Bruce Ferrell wrote:
> At the point where the phone line get's to your demarc the is supposed
> to ba a -2 to 3db reference point, sometimes called a -2 or -3 test
> level point (TLP).  So that milliwatt tone at that point should read in
> the range of -2 to -3 dbm.

Ok so since -3dB is 1/2 power we should be expecting a reading of 7422 in 
ztmonitor if it's a linear reading of the signal.

> If the milliwatt is arriving at the demarc at the nominal -2 to -3dbm
> and getting into the asterisk to be measured at 8dBm (+8dbm0), I'd say
> something is grossly mal-adjusted.  You're seeing 8db of gain!

No, he was saying he had to set his rxgain to 8 in order to get a level of 
14844 (0dBm) in ztmonitor.

-A.
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[Asterisk-Users] SIP Registration --Giving up forever after very short network outage.

[Steve Gladden]

I'm looking for some help in how to keep asterisk from doing this.
If we loose Internet or routing to our upstream provider even for only a
few short minutes asterisk quickly gives up & never tries again.
I have to do a manual reload to get it to register with my
sip provider(s) again before incoming calls are accepted.

This is really bad as it causes us to loose the ability to get incoming
calls now & then.
Not at all what we want in a phone system.


How can I get asterisk to stop giving up so soon or better yet not give
up at all... this is after all a phone system... I would really like
it to register back in if the Internet goes down then comes back up 10
minutes later!

I'm running CVS-HEAD (about two weeks old)
Aug 24 19:08:13 NOTICE[7124]: chan_sip.c:4701 sip_reg_timeout:--
Registration for '[EMAIL PROTECTED]' timed out, trying
again (Attempt #8)
Aug 24 19:08:16 NOTICE[7124]: chan_sip.c:4701 sip_reg_timeout:--
Registration for '[EMAIL PROTECTED]' timed out, trying again
(Attempt #11)
Aug 24 19:08:16 NOTICE[7124]: chan_sip.c:4719 sip_reg_timeout:--
Giving up forever trying to register '[EMAIL PROTECTED]'


Thanks for your help !!!

Steve


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[Asterisk-Users] Record(filename:gsm) very quiet

[Karl S. Katzke]

When I make a recording by setting up an extension with a record() 
statement in my dialplan, the resulting recording plays VERY quietly 
when it's played back over a POTS line, even though it sounds fine in 
the office. Is there any way to increase the gain on in-office 
recordings? I read the wiki and didn't find anything ... is there a 
better way to make these recordings for someone who doesn't have any 
sound equipment?
When the gsm file is playing back, it sounds like it's doing some funky 
slience compression... so when I start to speak, it won't be loud at the 
beginning, but it will slowly get louder. Whenever I pause, the first 
word I say when I start again will be quiet. This is also a problem when 
recording voicemail greetings.
I know that my voice sounds fine when I'm calling out, it's just the 
greetings that are faint.


Any suggestions?

Thanks,
Karl Katzke


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Re: [Asterisk-Users] MeetMe Marked user?

[Doug Lytle]

[EMAIL PROTECTED] wrote:



On Aug 24, 2005, at 11:21 AM, Doug Lytle wrote:



Create an extension that the user to be marked knows about, maybe 
even have it authenticate, mark the user and drop them into the 
conference.


Doug




If the Marked user isn't the first to enter the channel, then how does 
the MeetMe app know to put all other

users on hold until Marked user arrives? This is still unclear to me.


Example:

meetme.conf

conf => 1000

extensions.conf

; ** Normal users enter the conference here **
exten => 4823,1,SetMusicOnHold(random)
exten => 4823,2,Meetme(|Msciw)
exten => 4823,3,Hangup()

; ** Extension to mark conference users*

exten => 4824,1,Authenticate(12345)
exten => 4824,2,Meetme(|Asci)
exten => 4824,3,Hangup()


Users using extension 4823 and entering conference 1000 will listen to 
hold music until the marked users enters.


Users using extension 4824 and entering a password of 12345 will be able 
to select conference 1000 as the marked user.


Doug

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RE: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

[Alfredo J. Fabretti]

I have several production systems with tdm400p.

Do you have any IRQ conflict? cat /proc/interrupts

Did you try changing the pci slot?

Maybe the fxo module is damaged? did you try changing it to another 
place of the

card? I don't know if this can help, anyone?



Quoting canuck15 <[EMAIL PROTECTED]>:



Alfredo,

I tried a regular telco PSTN and a VoIP provider (webcall.ca using their
Nortel ATA connected to the TDM01B).  Both have very similar echo problems
using completely different wiring so I am quite convinced it has nothing to
do with the PSTN or wiring.

By the way, I sound just fine to the person on the other end.  They hear
absolutely no echo and say I sound crystal clear.

I also want to say that I am encouraged at the optimistic responses so far.
It tells me that there is hope if so many people feel this can work today
with existing hardware.

-Original Message-
From: Alfredo J. Fabretti [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 24, 2005 3:18 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

Try to use another land line and test the echo problem again.
Do you have any DSL service running in that line?




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[Asterisk-Users] Busy number signalling

[Eric Bishop]

Hi all,

Our Asterisk box sends calls outbound via either SIP (through our VoIP
provider) or an E1 PRI (directly connected via a TE410P). When we dial
a number that is engaged via our VoIP provider we get the following on
the Asterisk console (numbers and IP addresses changed to protect the
innocent):

   -- Called [EMAIL PROTECTED]
-- Got SIP response 486 "Busy here" back from 123.123.123.123
-- SIP/sip-outbound-af71 is busy
  == Everyone is busy/congested at this time

This is what we want as it then send the call to priority n+101 and we
can handle it any way we want from there. However if the outbound call
is made via the PRI to an enaged number it simply plays an enaged
signal to the caller and never progresses to priority n+101.

Anyone have any suggestions?
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Re: [Asterisk-Users] Someone can explain alerts in zap channels?

[Dave Cotton]

On Wed, 2005-08-24 at 16:43 -0700, Carlos Trallero wrote:
> Hi,
> 
>  Can someone explain me the meaning of a red alarm in
> a zap channel. Is there anyway of removing it?

Plugging a cable into it often helps.


-- 
Dave Cotton <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

[canuck15]

 
Alfredo,

I tried a regular telco PSTN and a VoIP provider (webcall.ca using their
Nortel ATA connected to the TDM01B).  Both have very similar echo problems
using completely different wiring so I am quite convinced it has nothing to
do with the PSTN or wiring.

By the way, I sound just fine to the person on the other end.  They hear
absolutely no echo and say I sound crystal clear. 

I also want to say that I am encouraged at the optimistic responses so far.
It tells me that there is hope if so many people feel this can work today
with existing hardware.

-Original Message-
From: Alfredo J. Fabretti [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 24, 2005 3:18 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

Try to use another land line and test the echo problem again.
Do you have any DSL service running in that line?


Quoting canuck15 <[EMAIL PROTECTED]>:

>
> I came into this with my eyes wide open.  I have read ABSOLUTELY 
> EVERYTHING there is to be found on the net about avoiding echo 
> problems BEFORE I even attempted to create a production system.  Since 
> lots of people are apparently using this in production environments 
> now I just assumed that echo IS avoidable.
>
> As others have recommended, I created a test system with the proposed 
> production parts.  I bought a couple different SIP phones to try and a 
> Digium TDM01B card.  I am using an older PIII 1Ghz system with 
> 815chipset (PCI Rev2.2) with 256MB for my test system.  The only thing 
> that will be different on a production system is that I will be using 
> a newer chipset PC with faster processor and 512MB.  Probably Intel 
> 7505, 7210, or 7211 chipsets which seem to be the most compatible with
Asterisk.
>
> My problem is that I cannot eliminate echo no matter what I try.  I 
> seriously doubt that a newer chipset faster PC with more memory will
> eliminate or even reduce my echo problems based on what I have read.   I
am
> not about to drop more cash to try and find out.  Essentially, my 
> findings are that Asterisk is NOT production capable for my 
> configuration which is via FXO and PSTN.  That is probably THE most 
> common configuration so if it is not production capable like that it 
> isn't production capable period as far as I'm concerned.  What a
disappointment :(.
>
> Unless I am missing something I am sure that many many people with a 
> similar configuration in a production environment have the same 
> problem.  Perhaps they are just living with it??  For me it is just as 
> unacceptable on an Asterisk system as it is on a traditional PBX.
> Some calls are ok and some are not.  No correlation to local, long 
> distance, time of day.  There always seems to be some echo.  Sometimes 
> it is worse than other times.  Again, no correlation to local, long 
> distance, time of day.  Tried connecting to ATA adapter and using VoIP 
> provider instead to see if the telco was causing the problem.  That 
> did not change anything.  Still the same general echo problem
>
> The things I have tried include in no particular order and not limited 
> to
> are:
>
> *Buy latest TDM400P with latest FXO module *Ensure copper connection 
> to analog telco lines and telco are not causing problems including 
> running a separate shielded line to the demarc AND having the telco 
> guy come out and test the levels, impedance etc.
> *Adjust RX/TX levels as per Asterisk Wiki using the quick Ztmonitor 
> method and by using the detailed Ztmonitor method via a Telco 
> 102milliwatt test phone #.  The end result was RX=8.0, TX=-1.0.  Since 
> I still have echo problems I have tried all sort of other settings without
success.
> *After ALL of the above, try every possible combination of all of the 
> following on Asterisk v1.0.9: echocancel (off, on, 128, 256, 16, 32, 
> 64), echowhenbridged (on, off), echotraining (off, on, 800), Mark 2 
> (default, aggressive, CVS head developments, bugs.digium.com patches, 
> adjust threshold level as per wiki etc. etc.) *Make sure echotraining 
> line is before FXO channel assignment in zapata.conf file *Run fxotune 
> which did not find a need to adjust the FXO levels
> (1=0,0,0,0,0,0,0,0)
>
> Based on all the above testing the best settings were pretty much in 
> line with what most people are finding.
> echocancel=on. echowhenbridged=on, echotraining=800, Mark 2 echo 
> canceller, aggressive cancellation OFF, bugs.digium.com #2820 patch,
RX=8.0, TX=-1.0.
>
> Still have echo.  Aggressive mode helps a bit but then the other 
> persons voice get's cut off a lot especially when I talk and the 
> cutting in and out of the canceller is more noticeable and 
> objectionable in general than if Aggressive is turned off.
>
> I have two SIP phones.  An Aastra 9133i and a Grandstream GXP2000.  
> Echo problem is the same on both phones.
>
>
> I am located within a metropolitan area in Canada.
>
> Any comments and/or suggestions would be grea

[Asterisk-Users] Channel ooh323c and DTMF with Call Manager

[Dan Austin]

Title: Channel ooh323c and DTMF with Call Manager






I had a need to rebuild my testing Asterisk system and thought I would see how

easy the new H.323 channel was to setup.


It turns out it is extremely easy, much nicer than dealing with pwlib and open_h323.


The only issue I have found is that version 0.0.2, the most recent I can find, only seems

to support rfc2833 for DTMF.  Cisco Call Manager does not seem to support rfc2833

on their H.323 trunks, so while the new channel was easier to setup, I cannot get much

use out of it.


Of course I may be missing something.  Has anyone tested the new channel with

Cisco Call Manager?


Thanks,

Dan



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[Asterisk-Users] Someone can explain alerts in zap channels?

[Carlos Trallero]

Hi,

 Can someone explain me the meaning of a red alarm in
a zap channel. Is there anyway of removing it?

 thanks
 Carlos


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[Asterisk-Users] Motherboards and IRQs

[jennyw]

Someone mentioned earlier (I can't find the message now) that they had a 
motherboard that allowed you to change IRQ assignments in BIOS. Does 
anyone happen to know how to identify motherboards that can do this? I'm 
going to put together a new machine now and I'm having trouble picking a 
motherboard for it (ordering from Dell or other online vendor is not an 
option, since I need this in the next couple days).


This is to provide another avenue for avoiding IRQ conflicts with the 
Digium TDM400Ps.


Thanks!

Jen

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Re: [Asterisk-Users] Asterisk Zaptel Leading Zero Problem With TE110P

[Christopher Mylonas]

What's the situation?

If you have asterisk coming off another pbx with a tie line, then you must set 
the variable below.

Try in zapata.conf
prilocaldialplan=local;

If you have it directly connected to the trunk, check your nationalprefix=0;
or something relevant to your actual country's settings.

Good luck

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Re: [Asterisk-Users] chan_capi, modprobe: Can't locate module capifs, ERROR: fopen(/etc/capi.conf, r)

[Goran Dj.]

> But, now I cannot start chan_capi.so:
> WARNING[802]: chan_capi.c:3106 load_module: CAPI not installed, CAPI
> disabled!
> 
> from tty:
> capiinit
> ERROR: cannot open /dev/capi20 nor /dev/isdn/capi20 - No such file or
> directory (2)
> 
> capiinfo
> capi not installed - No such file or directory (2)
> 
> capiinit show
> ERROR: fopen(/etc/capi.conf,r) failed - No such file or directory (2)
> ERROR: no cards configured in /etc/capi.conf


I resolved missing /dev/capi20 with shell script "makedev-capi.sh"
but, now, when starting capiinit:

modprobe: Can't locate module capifs
modprobe: Can't locate module capifs
WARNING: filesystem capifs not available
ERROR: fopen(/etc/capi.conf,r) failed - No such file or directory (2)



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RE: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

[canuck15]

Wiley,
 
The 
very first thing I checked was for IRQ problems.  I apoligize for 
forgetting to mention that.
 
The only thing I found in the Google search you 
suggested is a thread from January 2004 suggesting that businesses should ONLY 
use a T1 card with a channel bank instead of X100P cards.  I understand the 
"don't use X100P" part but I just assumed the TDM400P with echo cancellation is 
working fine now.
 
I am using a WRT54G (switch).  I am NOT connecting 
via wireless.  I have no traffic to QoS.  Just VoIP and a WRT54G 
switch is quite capable of that as far as I know.  No 
hubs!
 
I pretty much just use 
AAH now for it's ease of install but I have rolled my own in the 
past.  The echo has persisted through AAH v1.3, 1.4, 1.5.  I have 
recently (yesterday) installed the latest Asterisk and Zaptel CVS 
head development tree just to see if it made a difference and it DID 
NOT!
 
 
 
 



From: Wiley Siler 
[mailto:[EMAIL PROTECTED] Sent: Wednesday, August 24, 2005 
2:00 PMTo: Asterisk Users Mailing List - Non-Commercial 
DiscussionSubject: RE: [Asterisk-Users] Will Echo problems EVER be 
solved, I'm scared

Just because you cannot get it to work does not mean that 
IT does not work.  
 
Just using the right motherboard is not enough.  Did 
you check for IRQ problems?  You don't mention whether you have checked for 
this.
Look for a thread called "Asterisk-Users Small office 
setupusing analog lines w Asterisk" in the archive via 
Google.
use site:lists.digium.com
Try all the things listed in that 
thread.
 
Do you have a network that is capable of VoIP?  Are 
you using hubs when you should be using switches?
There is a major difference and hubs WILL NOT work reliably 
with VoIP.
Are you using QoS on your switches if you have lots of 
network traffic?
 
If you are using your own Distro and installing from 
scratch, try to use Asterisk at Home just to see if you still have the same 
problem.
 
I am putting my money on an IRQ issue 
myself.
 
W
 
 
 
 
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
canuck15Sent: Wednesday, August 24, 2005 1:38 PMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Will Echo 
problems EVER be solved, I'm scared

 
I came into this 
with my eyes wide open.  I have read ABSOLUTELY EVERYTHING there is to be 
found on the net about avoiding echo problems BEFORE I even attempted to create 
a production system.  Since lots of people are apparently using this in 
production environments now I just assumed that echo IS avoidable.  

 
As others have 
recommended, I created a test system with the proposed production parts.  I 
bought a couple different SIP phones to try and a Digium TDM01B card.  I am 
using an older PIII 1Ghz system with 815chipset (PCI Rev2.2) with 256MB for 
my test system.  The only thing that will be different on a production 
system is that I will be using a newer chipset PC with faster processor and 
512MB.  Probably Intel 7505, 7210, or 7211 chipsets which seem to be 
the most compatible with Asterisk.  
 
My problem is that I 
cannot eliminate echo no matter what I try.  I seriously doubt that a newer 
chipset faster PC with more memory will eliminate or even reduce my echo 
problems based on what I have read.   I am not about to drop more 
cash to try and find out.  Essentially, my findings are that Asterisk 
is NOT production capable for my configuration which is via FXO and PSTN.  
That is probably THE most common configuration so if it is not production 
capable like that it isn't production capable period as far as I'm 
concerned.  What a disappointment :(.  
 
Unless I am missing 
something I am sure that many many people with a similar configuration in a 
production environment have the same problem.  Perhaps they are just living 
with it??  For me it is just as unacceptable on an Asterisk system as it is 
on a traditional PBX.  Some calls are ok and some are not.  No 
correlation to local, long distance, time of day.  There always seems to be 
some echo.  Sometimes it is worse than other times.  Again, no 
correlation to local, long distance, time of day.  Tried connecting to ATA 
adapter and using VoIP provider instead to see if the telco was causing the 
problem.  That did not change anything.  Still the same general echo 
problem
 
The things I have 
tried include in no particular order and not limited to 
are:
 
*Buy latest TDM400P 
with latest FXO module
*Ensure copper 
connection to analog telco lines and telco are not causing problems including 
running a separate shielded line to the demarc AND having the telco guy come out 
and test the levels, impedance etc.
*Adjust RX/TX levels 
as per Asterisk Wiki using the quick Ztmonitor method and by using the detailed 
Ztmonitor method via a Telco 102milliwatt test phone #.  The end result was 
RX=8.0, TX=-1.0.  Since I still have echo problems I have tried all sort of 
other settings without success.
*After ALL of the 
above, try every possible combination of 

Re: [Asterisk-Users] chan_capi, cannot open /dev/capi20, no cards configured in /etc/capi.conf

[Goran Dj.]

> wget
>
ftp://ftp.isdn4linux.de/pub/isdn4linux/CVS-Snapshots/isdn4k-utils-CVS-2005-08-21.tar.bz2
>
> tar xvjf  isdn4k-utils-CVS-2005-08-21.tar.bz2
> cd isdn4k*
> cd capi20
> ./configure
> make
> make install
>
> that's all
>
> Sergio


Ok. Thanks. It's working, and I compiled successfully chan_capi-0.5.3
(because 0.5.4 producing some error).

But, now I cannot start chan_capi.so:
WARNING[802]: chan_capi.c:3106 load_module: CAPI not installed, CAPI
disabled!

from tty:
capiinit
ERROR: cannot open /dev/capi20 nor /dev/isdn/capi20 - No such file or
directory (2)

capiinfo
capi not installed - No such file or directory (2)

capiinit show
ERROR: fopen(/etc/capi.conf,r) failed - No such file or directory (2)
ERROR: no cards configured in /etc/capi.conf

So, whats happening? What is responsible for making /dev/capi20, and how
to make /etc/capi.conf?



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[Asterisk-Users] dingotel - connect Asterisk to 2-way radio?

[Shawn Rutledge]

So has anybody got one of these?  

http://www.amazon.com/exec/obidos/ASIN/B0007LQQUK/qid%3D1106972010/sr%3D11-1/ref%3Dsr%5F11%5F1/102-1529886-6420131

I'm thinking that it should be possible to connect it directly to an
Asterisk box and not use their software, as long as there was Linux
support for the USB dongle.  Maybe it just looks like a standard USB
audio device?  But there has to be an extra ring on the connector to
"key up" the radio so maybe to support that, the usual USB audio
driver would need some customization.  Just wondering if anybody has
tried it in Linux.

If you could do that, then Asterisk could listen for DTMF tones right?
 So you could use it from any 2-way radio which has a keypad, such as
a ham 2-meter rig.  (Or hack a keypad into your FRS or CB radio.)  Or,
Asterisk could be tied to a speech recognition system (like their
software has) but that is more complex.

Another aspect is the logic that is required to decide when to
transmit and when to listen.  You'd need to recognize the presence or
absence of human voice from both the VOIP end and the radio end, and
try to balance the two people's rights to monopolize the conversation.
 The limitation is that the radio is not full-duplex, like a repeater,
so when the person on the voip end is talking, and the radio whcih is
attached to the dingotel is transmitting that signal, it is not
listening, and nothing that the person on the remote radio end says is
going to get through.  It could be improved by actually using a
full-duplex radio though.
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[Asterisk-Users] FW: [Asterisk-Users, Andrew] Will Echo problems EVER be solved, I'm scared

[canuck15]

 
Thank you for the suggestions Andrew.  I have not come across some of them
before and will give them a shot.  Based on my reading, changing the
motherboard should have minimal impact unless that motherboard and the
TDM400P don't get along (aka. IRQ sharing).  I have disabled everything that
is not needed and I do not believe I have any IRQ problems and I am NEVER
wrong ;).  Calls are crisp and clear.  .  No snap, crackle, pop.  It would
be a beautiful thing if not for the echo.

To get the RX/TX levels, run "ztmonitor 1 -vv", dial a telco 1004hz 0dbm
test phone # and set the quantitative RX number to around 14500.  With 2
lines (which I don't have) you test the TX level by looping out to the other
PSTN.  Without a second line you do the simple ztmonitor test for the TX
levels.
http://www.voip-info.org/tiki-index.php?page=Asterisk+zapata+gain+adjustment

I was MOST DEFINITELY NOT wildly changing settings.  It would require a
whole book to explain properly what I did but the end result was that I
pretty much covered every possible combination of settings.  I have read the
white papers and EVERYTHING else I could find on the web to determine the
most logical and proper way to go about this.  I was NOT approaching this
like some back yard six pack scientist.

I made a mistake when I used the word "levels" to describe what fxotune
does.  The bottom line is that it did not change anything.

My settings are pretty much default except where I stated otherwise.
Network is a Linksys WRT54g (ie. Switch).  Asterisk server on port 1,
GXP2000 on port 2, 9133i on port 3.

I have NO echo between SIP phones!

#cat /proc/pci
Bus  0, device   0, function  0:
Host bridge: Intel Corp. 82815 815 Chipset Host Bridge and Memory
Controller Hub (rev 2).
  Prefetchable 32 bit memory at 0xd000 [0xd3ff].
  Bus  0, device   1, function  0:
PCI bridge: Intel Corp. 82815 815 Chipset AGP Bridge (rev 2).
  Master Capable.  Latency=32.  Min Gnt=12.
  Bus  0, device  30, function  0:
PCI bridge: Intel Corp. 82801BA/CA/DB/EB PCI Bridge (rev 1).
  Master Capable.  No bursts.  Min Gnt=6.
  Bus  0, device  31, function  0:
ISA bridge: Intel Corp. 82801BA ISA Bridge (LPC) (rev 1).
  Bus  0, device  31, function  1:
IDE interface: Intel Corp. 82801BA IDE U100 (rev 1).
  I/O at 0xf000 [0xf00f].
  Bus  0, device  31, function  3:
SMBus: Intel Corp. 82801BA/BAM SMBus (rev 1).
  IRQ 11.
  I/O at 0x5000 [0x500f].
  Bus  1, device   0, function  0:
VGA compatible controller: ATI Technologies Inc Rage 128 RF/SG AGP (rev
0).
  IRQ 10.
  Master Capable.  Latency=32.  Min Gnt=8.
  Prefetchable 32 bit memory at 0xd400 [0xd7ff].
  I/O at 0x9000 [0x90ff].
  Non-prefetchable 32 bit memory at 0xd900 [0xd9003fff].
  Bus  2, device   1, function  0:
Unknown mass storage controller: PCI device 1095:3124 (CMD Technology
Inc) (rev 1).
  IRQ 11.
  Master Capable.  Latency=32.
  Non-prefetchable 64 bit memory at 0xdb008000 [0xdb00807f].
  Non-prefetchable 64 bit memory at 0xdb00 [0xdb007fff].
  I/O at 0xa000 [0xa00f].
  Bus  2, device   2, function  0:
Communication controller: Tiger Jet Network Inc. Intel 537 (rev 0).
  IRQ 5.
  Master Capable.  Latency=32.  Min Gnt=1.Max Lat=128.
  I/O at 0xa400 [0xa4ff].
  Non-prefetchable 32 bit memory at 0xdb009000 [0xdb009fff].
  Bus  2, device   4, function  0:
Ethernet controller: Realtek Semiconductor Co., Ltd.
RTL-8139/8139C/8139C+ (rev 16).
  IRQ 10.
  Master Capable.  Latency=32.  Min Gnt=32.Max Lat=64.
  I/O at 0xa800 [0xa8ff].
  Non-prefetchable 32 bit memory at 0xdb00a000 [0xdb00a0ff].


#cat /proc/interrupts
 CPU0
  0:1290132  XT-PIC  timer
  1:  4  XT-PIC  keyboard
  2:  0  XT-PIC  cascade
  5:   12878920  XT-PIC  wctdm
  8:  1  XT-PIC  rtc
 10:  33866  XT-PIC  eth0
 12: 41  XT-PIC  PS/2 Mouse
 14:  17345  XT-PIC  ide0
 15: 60  XT-PIC  ide1
NMI:  0
ERR:  0



-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 24, 2005 1:54 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

On Wednesday 24 August 2005 16:37, canuck15 wrote:
> As others have recommended, I created a test system with the proposed 
> production parts.  I bought a couple different SIP phones to try and a 
> Digium TDM01B card.  I am using an older PIII 1Ghz system with 
> 815chipset (PCI Rev2.2) with 256MB for my test system.  The only thing 
> that will be different on a production system is that I will be using 
> a newer chipset PC with faster processor and 512MB.  Probably Intel 
> 7505, 7210, or 7211 chipsets which seem to be the most compatible with
Asterisk.

So in other words, everything will be changing on your production system.
No

Re: [Asterisk-Users] call parking timeout

[Eric Wieling aka ManxPower]

Brian May wrote:

On Wed, Aug 24, 2005 at 09:04:53AM -0500, Eric Wieling aka ManxPower wrote:


You looked at the features.conf.sample file?



Yes.

I don't see how that helps, at least in my version.

There is a parameter to change the timeout time, but I don't want to
change the time, I just want to change the behaviour when this timeout
is exceeded and the default behaviour doesn't work.

Or are you talking about some CVS-only feature here?



In 1.0.x parked calls will timeout to exten => s,1,Whatever in the 
correct context.  I don't know how CVS-HEAD handles it, but I do 
remember a discussion about CVS-HEAD doing the Right Thing, which is to 
timeout the call back to the extension that parked it.  I assumed that 
the behavour would be configurable, but I guess not.


--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120

r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care and don't insert
this by default into all your dial statements as you are killing call
progress information for the user. Really, you almost certainly do not
want to use this. Asterisk will generate ring tones automatically where
it is appropriate to do so. "r" makes it go the next step and
additionally generate ring tones where it is probably not appropriate to
do so.

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Re: [Asterisk-Users] call parking timeout

[Brian May]

On Wed, Aug 24, 2005 at 09:04:53AM -0500, Eric Wieling aka ManxPower wrote:
> You looked at the features.conf.sample file?

Yes.

I don't see how that helps, at least in my version.

There is a parameter to change the timeout time, but I don't want to
change the time, I just want to change the behaviour when this timeout
is exceeded and the default behaviour doesn't work.

Or are you talking about some CVS-only feature here?
-- 
Brian May <[EMAIL PROTECTED]>
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Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

[Alfredo J. Fabretti]

Try to use another land line and test the echo problem again.
Do you have any DSL service running in that line?


Quoting canuck15 <[EMAIL PROTECTED]>:



I came into this with my eyes wide open.  I have read ABSOLUTELY EVERYTHING
there is to be found on the net about avoiding echo problems BEFORE I even
attempted to create a production system.  Since lots of people are
apparently using this in production environments now I just assumed that
echo IS avoidable.

As others have recommended, I created a test system with the proposed
production parts.  I bought a couple different SIP phones to try and a
Digium TDM01B card.  I am using an older PIII 1Ghz system with 815chipset
(PCI Rev2.2) with 256MB for my test system.  The only thing that will be
different on a production system is that I will be using a newer chipset PC
with faster processor and 512MB.  Probably Intel 7505, 7210, or 7211
chipsets which seem to be the most compatible with Asterisk.

My problem is that I cannot eliminate echo no matter what I try.  I
seriously doubt that a newer chipset faster PC with more memory will
eliminate or even reduce my echo problems based on what I have read.   I am
not about to drop more cash to try and find out.  Essentially, my findings
are that Asterisk is NOT production capable for my configuration which is
via FXO and PSTN.  That is probably THE most common configuration so if it
is not production capable like that it isn't production capable period as
far as I'm concerned.  What a disappointment :(.

Unless I am missing something I am sure that many many people with a similar
configuration in a production environment have the same problem.  Perhaps
they are just living with it??  For me it is just as unacceptable on an
Asterisk system as it is on a traditional PBX.  Some calls are ok and some
are not.  No correlation to local, long distance, time of day.  There always
seems to be some echo.  Sometimes it is worse than other times.  Again, no
correlation to local, long distance, time of day.  Tried connecting to ATA
adapter and using VoIP provider instead to see if the telco was causing the
problem.  That did not change anything.  Still the same general echo problem

The things I have tried include in no particular order and not limited to
are:

*Buy latest TDM400P with latest FXO module
*Ensure copper connection to analog telco lines and telco are not causing
problems including running a separate shielded line to the demarc AND having
the telco guy come out and test the levels, impedance etc.
*Adjust RX/TX levels as per Asterisk Wiki using the quick Ztmonitor method
and by using the detailed Ztmonitor method via a Telco 102milliwatt test
phone #.  The end result was RX=8.0, TX=-1.0.  Since I still have echo
problems I have tried all sort of other settings without success.
*After ALL of the above, try every possible combination of all of the
following on Asterisk v1.0.9: echocancel (off, on, 128, 256, 16, 32, 64),
echowhenbridged (on, off), echotraining (off, on, 800), Mark 2 (default,
aggressive, CVS head developments, bugs.digium.com patches, adjust threshold
level as per wiki etc. etc.)
*Make sure echotraining line is before FXO channel assignment in zapata.conf
file
*Run fxotune which did not find a need to adjust the FXO levels
(1=0,0,0,0,0,0,0,0)

Based on all the above testing the best settings were pretty much in line
with what most people are finding.
echocancel=on. echowhenbridged=on, echotraining=800, Mark 2 echo canceller,
aggressive cancellation OFF, bugs.digium.com #2820 patch, RX=8.0, TX=-1.0.

Still have echo.  Aggressive mode helps a bit but then the other persons
voice get's cut off a lot especially when I talk and the cutting in and out
of the canceller is more noticeable and objectionable in general than if
Aggressive is turned off.

I have two SIP phones.  An Aastra 9133i and a Grandstream GXP2000.  Echo
problem is the same on both phones.


I am located within a metropolitan area in Canada.

Any comments and/or suggestions would be greatly appreciated as I am pretty
much out of ideas and ready to give up on Asterisk as a suitable traditional
small business phone system replacement.






--
Alfredo J. Fabretti
IPFLOW :: La inteligencia en sus comunicaciones
Argentina: (5411) 4294-8897
USA: (1) 914 301 8268
www.ip-flow.com.ar

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RE: [Asterisk-Users] fedora core 3 kernel source - could someone throw the dog a bone!

[Damon Estep]

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Harald Holzer
> Sent: Wednesday, August 24, 2005 3:56 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] fedora core 3 kernel source - could
someone
> throw the dog a bone!
> 
> > I know this is a question with an obvious answer to some, but I am
not
> > one of them.
> >
> > Installed FC3, but this time I decide to update since my ISOs are a
bit
> > old, so typical yum update
> >
> > Downloaded the FC3 SRPM for my kernel 2.6.12...
> >
> > Installed the SRPM package
> > Ran rpmbuild -bp -target=i686 kernel-2.6.spec
> 
> why to recompile the kernel package ?

Because I don't know any better?

Do I even need the kernel source RPM or just kernel-smp-devel?

When making zaptel;

Make clean
Make linux26 <-- is this still required in current CVS head?
Make install

> 
> >
> > Tried to build zaptel
> >
> > - error; You do not appear to have the sources for the
> > 2.6.12-1.1372_FC3smp kernel installed.
> 
> yum install kernel-smp-devel
> ln -s /lib/modules/`uname -r`/build/ /usr/src/linux-2.6
> 
> should help ;-)
> 
Yes it did, and I thank you!

Damon
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Re: [Asterisk-Users] fedora core 3 kernel source - could someone throw the dog a bone!

[Dave Cotton]

On Wed, 2005-08-24 at 23:56 +0200, Harald Holzer wrote:

The OP said
> > I know this is a question with an obvious answer to some, but I am not
> > one of them.
> >
> > Installed FC3, but this time I decide to update since my ISOs are a bit
> > old, so typical yum update
> >
> > Downloaded the FC3 SRPM for my kernel 2.6.12...
> >
> > Installed the SRPM package
> > Ran rpmbuild -bp -target=i686 kernel-2.6.spec
> 
> why to recompile the kernel package ?
> 
> >
> > Tried to build zaptel
> >
> > - error; You do not appear to have the sources for the
> > 2.6.12-1.1372_FC3smp kernel installed.
> 
> yum install kernel-smp-devel
> ln -s /lib/modules/`uname -r`/build/ /usr/src/linux-2.6
> 
> should help ;-)

Better to look at where /lib/modules/`uname -r`/build/ is actually
pointing to, you may be very surprised.  You did an rpmbuild ther's the
clue.

-- 
Dave Cotton <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] NAT and SIP.conf update.

[razza]

I did write a script to do this but blew away my box by accident
yesterday :o( if nothing exists I'll start googling shell commands again
;o)
Although when installing mandrake again today, I noticed a package
called wine which might be interesting.
Cheers

%<-- SNIP -->%

On Wednesday 24 Aug 2005 09:44, razza wrote:
> I have a standard BT home DSL, which means I cannot have a static IP 
> address, therefore i'm forced to use NAT, I subscribe to a DDNS 
> service and have written a VB app which polls the router every 10 
> seconds and updates the DDNS if appropriate.

Ditch your ISP and go with one who will give you a static IP.
B

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[Asterisk-Users] ANI2 AKA Info Digits not supported?

[Steve Edwards]

I'm not receiving ANI2 (info digits) on my SBC PRI's.

SBC said they're sending them.

I called Digium support and was told it is not supported.

Is anybody receiving ANI2 on a PRI?

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline   [EMAIL PROTECTED]Fax: +1-760-731-3000
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Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

[Michael D Schelin]

The Asterisk Software is not the problem.  I'm thinking and I could be
wrong that your having a total line balance mismatch with the card your
using.  Check the line impedance and the card's.   Most people using
Asterisk don't have that much echo. Anyway It would be nice to see a
manual Hybrid adjustment on analog cards.

Don't give up. 



canuck15 wrote:

  
  
   
  I
came into this with my eyes wide open.  I have read ABSOLUTELY
EVERYTHING there is to be found on the net about avoiding echo problems
BEFORE I even attempted to create a production system.  Since lots of
people are apparently using this in production environments now I just
assumed that echo IS avoidable.  
   
  As
others have recommended, I created a test system with the proposed
production parts.  I bought a couple different SIP phones to try and a
Digium TDM01B card.  I am using an older PIII 1Ghz system with
815chipset (PCI Rev2.2) with 256MB for my test system.  The only thing
that will be different on a production system is that I will be using a
newer chipset PC with faster processor and 512MB.  Probably Intel 7505,
7210, or 7211 chipsets which seem to be the most compatible with
Asterisk.  
   
  My
problem is that I cannot eliminate echo no matter what I try.  I
seriously doubt that a newer chipset faster PC with more memory will
eliminate or even reduce my echo problems based on what I have
read.   I am not about to drop more cash to try and find out. 
Essentially, my findings are that Asterisk is NOT production capable
for my configuration which is via FXO and PSTN.  That is probably THE
most common configuration so if it is not production capable like that
it isn't production capable period as far as I'm concerned.  What a
disappointment :(.  
   
  Unless
I am missing something I am sure that many many people with a similar
configuration in a production environment have the same problem. 
Perhaps they are just living with it??  For me it is just as
unacceptable on an Asterisk system as it is on a traditional PBX.  Some
calls are ok and some are not.  No correlation to local, long distance,
time of day.  There always seems to be some echo.  Sometimes it is
worse than other times.  Again, no correlation to local, long distance,
time of day.  Tried connecting to ATA adapter and using VoIP provider
instead to see if the telco was causing the problem.  That did not
change anything.  Still the same general echo problem
   
  The
things I have tried include in no particular order and not limited to
are:
   
  *Buy
latest TDM400P with latest FXO module
  *Ensure copper
connection to analog telco lines and telco are not causing problems
including running a separate shielded line to the demarc AND having the
telco guy come out and test the levels, impedance etc.
  *Adjust
RX/TX levels as per Asterisk Wiki using the quick Ztmonitor method and
by using the detailed Ztmonitor method via a Telco 102milliwatt test
phone #.  The end result was RX=8.0, TX=-1.0.  Since I still have echo
problems I have tried all sort of other settings without success.
  *After
ALL of the above, try every possible combination of all of the
following on Asterisk v1.0.9: echocancel (off, on, 128, 256, 16, 32,
64), echowhenbridged (on, off), echotraining (off, on, 800), Mark
2 (default, aggressive, CVS head developments, bugs.digium.com patches,
adjust threshold level as per wiki etc. etc.)
  *Make
sure echotraining line is before FXO channel assignment in zapata.conf
file
  *Run
fxotune which did not find a need to adjust the FXO levels
(1=0,0,0,0,0,0,0,0)
   
  Based
on all the above testing the best settings were pretty much in line
with what most people are finding. 
  echocancel=on.
echowhenbridged=on, echotraining=800, Mark 2 echo canceller, aggressive
cancellation OFF, bugs.digium.com #2820 patch, RX=8.0, TX=-1.0.
   
  Still
have echo.  Aggressive mode helps a bit but then the other persons
voice get's cut off a lot especially when I talk and the cutting in and
out of the canceller is more noticeable and objectionable in general
than if Aggressive is turned off.
   
  I have
two SIP phones.  An Aastra 9133i and a Grandstream GXP2000.  Echo
problem is the same on both phones. 
   
   
  I
am located within a metropolitan area in Canada.
   
  Any comments and/or suggestions would be greatly
appreciated as I am pretty much out of ideas and ready to give up on
Asterisk as a suitable traditional small business phone system
replacement.
    
  

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Re: [Asterisk-Users] fedora core 3 kernel source - could someone throw the dog a bone!

[Harald Holzer]

> I know this is a question with an obvious answer to some, but I am not
> one of them.
>
> Installed FC3, but this time I decide to update since my ISOs are a bit
> old, so typical yum update
>
> Downloaded the FC3 SRPM for my kernel 2.6.12...
>
> Installed the SRPM package
> Ran rpmbuild -bp -target=i686 kernel-2.6.spec

why to recompile the kernel package ?

>
> Tried to build zaptel
>
> - error; You do not appear to have the sources for the
> 2.6.12-1.1372_FC3smp kernel installed.

yum install kernel-smp-devel
ln -s /lib/modules/`uname -r`/build/ /usr/src/linux-2.6

should help ;-)


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Re: [Asterisk-Users] AGI + Ruby

[Mark Phillips]

YEah but the problem with pearls are that they come in different colours 
and are often of varying quality.


Black one (which are actually green) are the best.



Huddleston, Robert wrote:

Actually Perl is even better
 
 




-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
Mark Phillips

Sent: Wednesday, August 24, 2005 4:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AGI + Ruby

Y'see it? There it goes! Right over his head.


Huddleston, Robert wrote:

U joke - duh! 







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


Of Innocent 


Evil
Sent: Wednesday, August 24, 2005 3:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AGI + Ruby

What IDE are you talking about?
Any URL would be helpful.

Thanks,






-Original Message-
From: [EMAIL PROTECTED]
Sent: Wed, 24 Aug 2005 15:16:18 -0400
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] AGI + Ruby

I think you might find amethyst much simpler and possibly


cheaper too. 




I believe the current IDE is 12.4K



Innocent Evil wrote:



I would like to write AGI script in Ruby Would anybody


please show



me right direction..


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--

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Randolph, NJ
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--

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[Asterisk-Users] [Asterisk-Dev] Job Opening - Release Engineer

[Paul Mahler]

Signate has an immediate opening for a qa/release engineer for our line of VoIP
telephony products. 

Release Engineer

Signate is rapidly growing and profitable. We are about to launch a new line of
telephone software products. That’s where you can come into the picture. 

You would support Signate's software development team by reviewing new and
changed code, tracking and auditing change histories, debugging build and
runtime problems, and maintaining a build process to support ongoing R&D and
regression and user/system level tests. As a Release Engineer you will have
primary responsibility for updating release branches in our source control
system, building and testing release binaries, and pushing releases to
production. You will design and document improvements to the integration /
build / test and release processes.

Our development team is distributed around the world, and you could be located
anywhere. If you have a passion for testing, are a quick learner,
self-motivated and capable of working independently as an integral part of a
team we’d like to talk to you!. 

Job Requirements: 

Minimum of three years' software QA and configuration management experience. 

Genuine enjoyment of SQA work. 

Strong knowledge of Internet technologies, mySQL, PHP and the Linux operating
system. Exposure to XML/XSL and JSP. 

Experience with c and Asterisk source code. 

Proficiency with software testing automation tools.
 
Ability to create effective test plans. 

Ability to prioritize problems in problem tracking software applications 

Experience with software configuration management systems / source code version
control systems. 

Must have excellent technical writing and communication skills, and strong
problem solving skills. 

Send your resume and salary requirements to Paul Mahler at [EMAIL PROTECTED]



Paul Mahler
www.signate.com
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Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

[Paul]

Has anyone tried this approach?

1) Install * on a PC(probably don't need much horsepower)

2) Setup a sipura spa-2000 ata so that it is not on the same lan you are 
troubleshooting. One way to do this is with a crossover cable to the 
above PC. Restrict both ata ports to ulaw only.


3) Port 1 of ata gets a good analog phone

4) Port 2 simulates a pots line. Run a quality short cable to the fxo 
you are testing.


This way you can also test things like caller ID without paying a telco.

I suppose you could also use the ata that comes with vonage and others 
to test an fxo. As long as you get good call quality it should work.



Wiley Siler wrote:

Just because you cannot get it to work does not mean that IT does not 
work. 
 
Just using the right motherboard is not enough.  Did you check for IRQ 
problems?  You don't mention whether you have checked for this.
Look for a thread called "Asterisk-Users Small office setupusing 
analog lines w Asterisk" in the archive via Google.

use site:lists.digium.com
Try all the things listed in that thread.
 
Do you have a network that is capable of VoIP?  Are you using hubs 
when you should be using switches?

There is a major difference and hubs WILL NOT work reliably with VoIP.
Are you using QoS on your switches if you have lots of network traffic?
 
If you are using your own Distro and installing from scratch, try to 
use Asterisk at Home just to see if you still have the same problem.
 
I am putting my money on an IRQ issue myself.
 
W
 
 
 
 
 



*From:* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *On Behalf Of *canuck15

*Sent:* Wednesday, August 24, 2005 1:38 PM
*To:* asterisk-users@lists.digium.com
*Subject:* [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

 
I came into this with my eyes wide open.  I have read ABSOLUTELY 
EVERYTHING there is to be found on the net about avoiding echo 
problems BEFORE I even attempted to create a production system.  Since 
lots of people are apparently using this in production environments 
now I just assumed that echo IS avoidable. 
 
As others have recommended, I created a test system with the proposed 
production parts.  I bought a couple different SIP phones to try and a 
Digium TDM01B card.  I am using an older PIII 1Ghz system with 
815chipset (PCI Rev2.2) with 256MB for my test system.  The only thing 
that will be different on a production system is that I will be using 
a newer chipset PC with faster processor and 512MB.  Probably Intel 
7505, 7210, or 7211 chipsets which seem to be the most compatible with 
Asterisk. 
 
My problem is that I cannot eliminate echo no matter what I try.  I 
seriously doubt that a newer chipset faster PC with more memory will 
eliminate or even reduce my echo problems based on what I have 
read.   I am not about to drop more cash to try and find out.  
Essentially, my findings are that Asterisk is NOT production capable 
for my configuration which is via FXO and PSTN.  That is probably THE 
most common configuration so if it is not production capable like that 
it isn't production capable period as far as I'm concerned.  What a 
disappointment :(. 
 
Unless I am missing something I am sure that many many people with a 
similar configuration in a production environment have the same 
problem.  Perhaps they are just living with it??  For me it is just as 
unacceptable on an Asterisk system as it is on a traditional PBX.  
Some calls are ok and some are not.  No correlation to local, long 
distance, time of day.  There always seems to be some echo.  Sometimes 
it is worse than other times.  Again, no correlation to local, long 
distance, time of day.  Tried connecting to ATA adapter and using VoIP 
provider instead to see if the telco was causing the problem.  That 
did not change anything.  Still the same general echo problem
 
The things I have tried include in no particular order and not limited 
to are:
 
*Buy latest TDM400P with latest FXO module
*Ensure copper connection to analog telco lines and telco are not 
causing problems including running a separate shielded line to the 
demarc AND having the telco guy come out and test the levels, 
impedance etc.
*Adjust RX/TX levels as per Asterisk Wiki using the quick Ztmonitor 
method and by using the detailed Ztmonitor method via a Telco 
102milliwatt test phone #.  The end result was RX=8.0, TX=-1.0.  Since 
I still have echo problems I have tried all sort of other settings 
without success.
*After ALL of the above, try every possible combination of all of the 
following on Asterisk v1.0.9: echocancel (off, on, 128, 256, 16, 32, 
64), echowhenbridged (on, off), echotraining (off, on, 800), Mark 
2 (default, aggressive, CVS head developments, bugs.digium.com 
patches, adjust threshold level as per wiki etc. etc.)
*Make sure echotraining line is before FXO channel assignment in 
zapata.conf file
*Run fxotune which did not find a need to adjust the FXO 

Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

[Bruce Ferrell]

OK comments on echo and levels.

I made a living doing this in a central office so take it for what it's 
worth.


Milliwatt is 0dbm0 or 0dbm at a 0 reference point.

At the point where the phone line get's to your demarc the is supposed 
to ba a -2 to 3db reference point, sometimes called a -2 or -3 test 
level point (TLP).  So that milliwatt tone at that point should read in 
the range of -2 to -3 dbm.


Voice BTW, is considered to be a nominal -15dbm0.

The digital stream of a T1/E1 is considered to be a 0 reference point. 
When I worked on telephone switches (NorTel DMS250) the entire switch, 
because it was all digital was considered to be a 0 TLP.


If the milliwatt is arriving at the demarc at the nominal -2 to -3dbm 
and getting into the asterisk to be measured at 8dBm (+8dbm0), I'd say 
something is grossly mal-adjusted.  You're seeing 8db of gain!


Fix that and your echo should go away.

P.S.

With that much gain, there is no echo cancellor that I know that can 
cope, hard or soft.


canuck15 wrote:
 
I came into this with my eyes wide open.  I have read ABSOLUTELY 
EVERYTHING there is to be found on the net about avoiding echo problems 
BEFORE I even attempted to create a production system.  Since lots of 
people are apparently using this in production environments now I just 
assumed that echo IS avoidable. 
 
As others have recommended, I created a test system with the proposed 
production parts.  I bought a couple different SIP phones to try and a 
Digium TDM01B card.  I am using an older PIII 1Ghz system with 
815chipset (PCI Rev2.2) with 256MB for my test system.  The only thing 
that will be different on a production system is that I will be using a 
newer chipset PC with faster processor and 512MB.  Probably Intel 7505, 
7210, or 7211 chipsets which seem to be the most compatible with Asterisk. 
 
My problem is that I cannot eliminate echo no matter what I try.  I 
seriously doubt that a newer chipset faster PC with more memory will 
eliminate or even reduce my echo problems based on what I have read.   I 
am not about to drop more cash to try and find out.  Essentially, my 
findings are that Asterisk is NOT production capable for my 
configuration which is via FXO and PSTN.  That is probably THE most 
common configuration so if it is not production capable like that 
it isn't production capable period as far as I'm concerned.  What a 
disappointment :(. 
 
Unless I am missing something I am sure that many many people with a 
similar configuration in a production environment have the same 
problem.  Perhaps they are just living with it??  For me it is just as 
unacceptable on an Asterisk system as it is on a traditional PBX.  Some 
calls are ok and some are not.  No correlation to local, long distance, 
time of day.  There always seems to be some echo.  Sometimes it is worse 
than other times.  Again, no correlation to local, long distance, time 
of day.  Tried connecting to ATA adapter and using VoIP provider instead 
to see if the telco was causing the problem.  That did not change 
anything.  Still the same general echo problem
 
The things I have tried include in no particular order and not limited 
to are:
 
*Buy latest TDM400P with latest FXO module
*Ensure copper connection to analog telco lines and telco are not 
causing problems including running a separate shielded line to the 
demarc AND having the telco guy come out and test the levels, impedance etc.
*Adjust RX/TX levels as per Asterisk Wiki using the quick Ztmonitor 
method and by using the detailed Ztmonitor method via a Telco 
102milliwatt test phone #.  The end result was RX=8.0, TX=-1.0.  Since I 
still have echo problems I have tried all sort of other settings without 
success.
*After ALL of the above, try every possible combination of all of the 
following on Asterisk v1.0.9: echocancel (off, on, 128, 256, 16, 32, 
64), echowhenbridged (on, off), echotraining (off, on, 800), Mark 
2 (default, aggressive, CVS head developments, bugs.digium.com patches, 
adjust threshold level as per wiki etc. etc.)
*Make sure echotraining line is before FXO channel assignment in 
zapata.conf file
*Run fxotune which did not find a need to adjust the FXO levels 
(1=0,0,0,0,0,0,0,0)
 
Based on all the above testing the best settings were pretty much in 
line with what most people are finding. 
echocancel=on. echowhenbridged=on, echotraining=800, Mark 2 echo 
canceller, aggressive cancellation OFF, bugs.digium.com #2820 patch, 
RX=8.0, TX=-1.0.
 
Still have echo.  Aggressive mode helps a bit but then the other persons 
voice get's cut off a lot especially when I talk and the cutting in and 
out of the canceller is more noticeable and objectionable in general 
than if Aggressive is turned off.
 
I have two SIP phones.  An Aastra 9133i and a Grandstream GXP2000.  Echo 
problem is the same on both phones. 
 
 
I am located within a metropolitan area in Canada.
 
Any comments and/or suggestions would be greatly appreciated as I am 
pretty muc

[Asterisk-Users] google talk sniff

[Jason p]

for anyone wanting to see what ports the voice connection runs on:

Internet Protocol, Src: 66.162.X.X (66.162.X.X), Dst: 192.168.1.21
(192.168.1.21)
Version: 4
Header length: 20 bytes
Differentiated Services Field: 0x80 (DSCP 0x20: Class Selector 4; ECN: 0x00)
Total Length: 116
Identification: 0x462e (17966)
Flags: 0x00
Fragment offset: 0
Time to live: 112
Protocol: UDP (0x11)
Header checksum: 0xf59b [correct]
Source: 66.162.X.X (66.162.X.X)
Destination: 192.168.1.21 (192.168.1.21)
User Datagram Protocol, Src Port: 37897 (37897), Dst Port: 1142 (1142)
Source port: 37897 (37897)
Destination port: 1142 (1142)
Length: 96
Checksum: 0xeaaf [correct]
Data (88 bytes)
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RE: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

[Wiley Siler]

Just because you cannot get it to work does not mean that 
IT does not work.  
 
Just using the right motherboard is not enough.  Did 
you check for IRQ problems?  You don't mention whether you have checked for 
this.
Look for a thread called "Asterisk-Users Small office 
setupusing analog lines w Asterisk" in the archive via 
Google.
use site:lists.digium.com
Try all the things listed in that 
thread.
 
Do you have a network that is capable of VoIP?  Are 
you using hubs when you should be using switches?
There is a major difference and hubs WILL NOT work reliably 
with VoIP.
Are you using QoS on your switches if you have lots of 
network traffic?
 
If you are using your own Distro and installing from 
scratch, try to use Asterisk at Home just to see if you still have the same 
problem.
 
I am putting my money on an IRQ issue 
myself.
 
W
 
 
 
 
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
canuck15Sent: Wednesday, August 24, 2005 1:38 PMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Will Echo 
problems EVER be solved, I'm scared

 
I came into this 
with my eyes wide open.  I have read ABSOLUTELY EVERYTHING there is to be 
found on the net about avoiding echo problems BEFORE I even attempted to create 
a production system.  Since lots of people are apparently using this in 
production environments now I just assumed that echo IS avoidable.  

 
As others have 
recommended, I created a test system with the proposed production parts.  I 
bought a couple different SIP phones to try and a Digium TDM01B card.  I am 
using an older PIII 1Ghz system with 815chipset (PCI Rev2.2) with 256MB for 
my test system.  The only thing that will be different on a production 
system is that I will be using a newer chipset PC with faster processor and 
512MB.  Probably Intel 7505, 7210, or 7211 chipsets which seem to be 
the most compatible with Asterisk.  
 
My problem is that I 
cannot eliminate echo no matter what I try.  I seriously doubt that a newer 
chipset faster PC with more memory will eliminate or even reduce my echo 
problems based on what I have read.   I am not about to drop more 
cash to try and find out.  Essentially, my findings are that Asterisk 
is NOT production capable for my configuration which is via FXO and PSTN.  
That is probably THE most common configuration so if it is not production 
capable like that it isn't production capable period as far as I'm 
concerned.  What a disappointment :(.  
 
Unless I am missing 
something I am sure that many many people with a similar configuration in a 
production environment have the same problem.  Perhaps they are just living 
with it??  For me it is just as unacceptable on an Asterisk system as it is 
on a traditional PBX.  Some calls are ok and some are not.  No 
correlation to local, long distance, time of day.  There always seems to be 
some echo.  Sometimes it is worse than other times.  Again, no 
correlation to local, long distance, time of day.  Tried connecting to ATA 
adapter and using VoIP provider instead to see if the telco was causing the 
problem.  That did not change anything.  Still the same general echo 
problem
 
The things I have 
tried include in no particular order and not limited to 
are:
 
*Buy latest TDM400P 
with latest FXO module
*Ensure copper 
connection to analog telco lines and telco are not causing problems including 
running a separate shielded line to the demarc AND having the telco guy come out 
and test the levels, impedance etc.
*Adjust RX/TX levels 
as per Asterisk Wiki using the quick Ztmonitor method and by using the detailed 
Ztmonitor method via a Telco 102milliwatt test phone #.  The end result was 
RX=8.0, TX=-1.0.  Since I still have echo problems I have tried all sort of 
other settings without success.
*After ALL of the 
above, try every possible combination of all of the following on Asterisk 
v1.0.9: echocancel (off, on, 128, 256, 16, 32, 64), echowhenbridged (on, off), 
echotraining (off, on, 800), Mark 2 (default, aggressive, CVS head 
developments, bugs.digium.com patches, adjust threshold level as per wiki etc. 
etc.)
*Make sure 
echotraining line is before FXO channel assignment in zapata.conf 
file
*Run fxotune which 
did not find a need to adjust the FXO levels 
(1=0,0,0,0,0,0,0,0)
 
Based on all the 
above testing the best settings were pretty much in line with what most people 
are finding. 
echocancel=on. 
echowhenbridged=on, echotraining=800, Mark 2 echo canceller, aggressive 
cancellation OFF, bugs.digium.com #2820 patch, RX=8.0, 
TX=-1.0.
 
Still have 
echo.  Aggressive mode helps a bit but then the other persons voice get's 
cut off a lot especially when I talk and the cutting in and out 
of the canceller is more noticeable and objectionable in general 
than if Aggressive is turned off.
 
I have two SIP 
phones.  An Aastra 9133i and a Grandstream GXP2000.  Echo problem is 
the same on both phones. 
 
 
I am located within 
a metropolitan area in Canada.
 
A

Re: [Asterisk-Users] chan_capi on slackware10? cannot compile :-( why?

[Sergio Chersovani]

Yes, but I dont have it on my Slackware10 CD'es. I don't have libcapi,
or isdn4... or anything with isdn or capi in their name. Where to find
libcapi20 (od devel...) for slackware?
 

wget 
ftp://ftp.isdn4linux.de/pub/isdn4linux/CVS-Snapshots/isdn4k-utils-CVS-2005-08-21.tar.bz2


tar xvjf  isdn4k-utils-CVS-2005-08-21.tar.bz2
cd isdn4k*
cd capi20
./configure
make
make install

that's all

Sergio
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Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

[Andrew Kohlsmith]

On Wednesday 24 August 2005 16:37, canuck15 wrote:
> As others have recommended, I created a test system with the proposed
> production parts.  I bought a couple different SIP phones to try and a
> Digium TDM01B card.  I am using an older PIII 1Ghz system with 815chipset
> (PCI Rev2.2) with 256MB for my test system.  The only thing that will be
> different on a production system is that I will be using a newer chipset PC
> with faster processor and 512MB.  Probably Intel 7505, 7210, or 7211
> chipsets which seem to be the most compatible with Asterisk.

So in other words, everything will be changing on your production system.  Not 
a good way to start.

> My problem is that I cannot eliminate echo no matter what I try.  I
> seriously doubt that a newer chipset faster PC with more memory will
> eliminate or even reduce my echo problems based on what I have read.   I am
> not about to drop more cash to try and find out.  Essentially, my findings
> are that Asterisk is NOT production capable for my configuration which is
> via FXO and PSTN.  That is probably THE most common configuration so if it
> is not production capable like that it isn't production capable period as
> far as I'm concerned.  What a disappointment :(.

Most of us don't have any trouble.

> *Buy latest TDM400P with latest FXO module
> *Ensure copper connection to analog telco lines and telco are not causing
> problems including running a separate shielded line to the demarc AND
> having the telco guy come out and test the levels, impedance etc.

I'd be damn curious to know what you got out of this -- most telco guys will 
do a basic metallic check, throw on a butt-set and say "yup, I got dialtone." 
-- hardly a real check but that's neither here nor there.  I'm also in Canada 
(1.5hrs from Toronto, ON) so I'm *really* curious who you got on the line to 
do a real line test with you.  I have resorted to buying my own telco test 
equipment off ebay and using that, even though our techs here are excellent.

> *Adjust RX/TX levels as per Asterisk Wiki using the quick Ztmonitor method
> and by using the detailed Ztmonitor method via a Telco 102milliwatt test
> phone #.  The end result was RX=8.0, TX=-1.0.  Since I still have echo
> problems I have tried all sort of other settings without success.

Ok good.  Can you detail exactly what you did to reach these numbers?  I'm 
curious.

> *After ALL of the above, try every possible combination of all of the
> following on Asterisk v1.0.9: echocancel (off, on, 128, 256, 16, 32, 64),
> echowhenbridged (on, off), echotraining (off, on, 800), Mark 2 (default,
> aggressive, CVS head developments, bugs.digium.com patches, adjust
> threshold level as per wiki etc. etc.)

I'd posted something earlier that basically says this:  Without measured, 
controlled tests, you're just pissing up a rope.  Wildly changing settings 
and hoping for the best does nothing but cost you time and energy.

> *Run fxotune which did not find a need to adjust the FXO levels
> (1=0,0,0,0,0,0,0,0)

fxotune doesn't adjust FXO levels, it adjusts a very simple FIR filter which 
is part of the DAA in the FXO module.  IMO it helps with audio quality but 
not much with echo.

> Still have echo.  Aggressive mode helps a bit but then the other persons
> voice get's cut off a lot especially when I talk and the cutting in and out
> of the canceller is more noticeable and objectionable in general than if
> Aggressive is turned off.

Agressive mode turns the phone line into a half-duplex environment.  When your 
voice energy is detected it mutes the receive audio.

> I have two SIP phones.  An Aastra 9133i and a Grandstream GXP2000.  Echo
> problem is the same on both phones.

Do you have echo between the two phones?  What about when calling out to a 
VOIP provider, dialing a DID you own that comes back in and hits the other 
phone?

> Any comments and/or suggestions would be greatly appreciated as I am pretty
> much out of ideas and ready to give up on Asterisk as a suitable
> traditional small business phone system replacement.

I haven't seen your zconfig.h nor your zaptel Makefile, and you didn't tell us 
anything about your network (network card, switch, etc.).

My general advice for zaptel is to do the following:
zaptel Makefile: underneath the comments about zconfig.h add
KFLAGS+=-march=pentium4 (or pentium3 or pentiumpro, use the exact proc)
CFLAGS+=-march=pentium4 (or pentium3 or pentiumpro, use the exact proc)

and in zconfig.h
- enable XLAW (optimize for small # of zap channels)
- enable MMX
- MARK2, no agressive mode.

Whenever I've done that my echo has largely disappeared.  

Have you also tried flipping tip and ring going into the TDM card?

-A.
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[Asterisk-Users] ASTCC and cdrs

[Il Neofita]

My installation of ASTCC does not update the cdrs tables .
It is a problem of ASTCC or it is a configuration problem?
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[Asterisk-Users] Error when answering CAPI

[Humberto Aicardi]

Hi,

   I've a Fritz card which was working fine, recently I changed 
hardware and my nightmare started. Now when I call someone through the 
chan_capi (0.3.5 or 0.4.0) it works fine but when I receive calls I 
always get hungup. Can someone please give some help? Here are the logs:


*CLI>
   -- CONNECT_IND ID=001 #0x LEN=0049
 Controller/PLCI/NCCI= 0x101
 CIPValue= 0x1
 CalledPartyNumber   = 1138121122
 CallingPartyNumber  = A<83>1181114000
 CalledPartySubaddress   = default
 CallingPartySubaddress  = default
 BC  = <80 90 a3>
 LLC = default
 HLC = default
 AdditionalInfo  = default

 == CONNECT_IND 
(PLCI=0x101,DID=1138121122,CID=1181114000,CIP=0x1,CONTROLLER=0x1)

   -- creating pipe for PLCI=0x101 msn = *
   -- INFO_IND ID=001 #0x0001 LEN=0026
 Controller/PLCI/NCCI= 0x101
 InfoNumber  = 0x70
 InfoElement = 1138121122

   -- INFO_IND ID=001 #0x0001 LEN=0026
 Controller/PLCI/NCCI= 0x101
 InfoNumber  = 0x70
 InfoElement = 1138121122

  > sent INFO_RESP (PLCI=0x101)
  > sent CONNECT_RESP for PLCI = 0x101
   -- INFO_IND ID=001 #0x0002 LEN=0016
 Controller/PLCI/NCCI= 0x101
 InfoNumber  = 0x18
 InfoElement = <89>

   -- INFO_IND ID=001 #0x0002 LEN=0016
 Controller/PLCI/NCCI= 0x101
 InfoNumber  = 0x18
 InfoElement = <89>

  > sent INFO_RESP (PLCI=0x101)
   -- DISCONNECT_IND ID=001 #0x0003 LEN=0014
 Controller/PLCI/NCCI= 0x101
 Reason  = 0x0

 == DISCONNECT_IND PLCI=0x101 REASON=0
  > sent DISCONNECT_RESP PLCI=0x101
   -- CAPI Hangingup
  > activehangingup
   -- removed pipe for PLCI = 0x101


capi.conf
===
;
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8

[interfaces]

; mode: ptmp (point-to-multipoint) or ptp (point-to-point)
isdnmode=ptp
; allow incoming calls to this list of MSNs, * == any
incomingmsn=*
; capi controller number
controller=1
; dialout group
group=1
; enable/disable software dtmf detection, recommended for AVM cards
softdtmf=1
; accountcode to use in CDRs
accountcode=
; context for incoming calls
context=from-pstn
; _VERY_PRIMITIVE_ echo suppression
;echosquelch=1
; EICON DIVA SERVER echo cancelation
;echocancel=yes
;echotail=64
; call group
;callgroup=1
; deflect incoming calls to 12345678 if all B channels are busy
;deflect=12345678
; number of concurrent calls on this controller (2 makes sense for 
single BRI)

devices => 2




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[Asterisk-Users] Re: Asterisk and MWI

[Joe McConnaughey]

Title: Message



Melissa -
 
I added the "fromuser=AnyName" to my sip.conf file stations and that, in 
fact, corrected the problem.  The MWI now works 
flawlessly.
 
I 
would recommend that Aastra/Sayson pursue this with the Asterisk team so 
that it is listed as a known issue or to have Asterisk patched to fix it.  
I will submit a bug on it as well.  I'm copying the Users Mailing List on 
this.
 
For 
your records, I'm running Asterisk HEAD as of 08/22/2005.  This is the 
very latest version of Asterisk from the CVS 
repository.
 
Please thank everyone for their hard work and fact finding.  
I think you have tremendous customer support!  It only took a couple of days to track and correct this 
issue.  I'm extremely pleased with your phones and your 
support!
 
Joe 
McConnaughey

  - Original Message - 
  From: 
  Melissa 
  Lee 
  To: [EMAIL PROTECTED] 
  Sent: Wednesday, August 24, 2005 4:05 
  PM
  Subject: FW: Asterisk and MWI 
  
  Joe
  
   
  Following is the findings from our expert. Please let 
  me know which version of Asterisk that you are using and please also forward 
  your extension.conf to me so that we can compare the files that work in 
  our test lab. Thanks. Melissa. 
   
  
  Currently there are two known 
  issues (both resolved) that have led customers to complain that MWI isn’t 
  working with 1.2.x
   
   
  The first problem is most likely 
  reported as MWI worked in 1.0.0.78 but doesn’t work with 1.2.x.  This is 
  because there was a bug in Asterisk versions prior to 1.0.4 that meant it 
  wasn’t compliant to RFC3842.  One of the issues also stopped the message 
  count working in 1.0.0.78, but at least the LED came on.   The 
  solution is to upgrade Asterisk to version 1.0.4 or later, the latest is 
  1.0.9
   
  The second problem is 
  characterised by the phone not responding to MWI messages from Asterisk (I 
  think 1.0.0.78 exhibits the same problem).  This is caused by an illegal 
  “From” header in the NOTIFY message from Asterisk:
   
  NOTIFY 
  sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP 
  192.168.0.101:5060;branch=z9hG4bK5a616fef;rportFrom: "asterisk" 
  ;tag=as09997aedTo: 
  Contact: 
  Call-ID: 
  [EMAIL PROTECTED]CSeq: 102 
  NOTIFYUser-Agent: Asterisk 
  PBXEvent: 
  message-summaryContent-Type: 
  application/simple-message-summaryContent-Length: 
  43
   
  Messages-Waiting: 
  yesVoice-Message: 
  3/0
   
   
  I don’t know if this is an 
  Asterisk bug or configuration problem, so if someone comes across this issue 
  can they gather the voicemail.conf, sip.conf and extension.conf files from the 
  asterisk server and version that is running.  In the meantime, Mani found 
  a workaround for this:-
   
  We have to tell our 
  customers to add following line
   
  fromuser=AnyName
   
  to each user setting 
  in the sip.conf on their asterisk server.
   
  
  
  
  
  Sayson 
  Technologies Ltd.
  210 - 1910 Quebec 
  St
  Vancouver, BC  
  V5T4K1
  Canada
  Phone: 
  604.730.1842
  Fax: 
  604.732.8726
   
  
  
  
  This email and any 
  files transmitted with it are confidential material. They are intended solely 
  for the use of the designated individual or entity to whom they are addressed. 
  If the reader of this message is not the intended recipient, you are hereby 
  notified that any dissemination, use, distribution or copying of this communication is strictly 
  prohibited and may be unlawful.If you have received this email in 
  error please notify [EMAIL PROTECTED] and permanently delete the e-mail and 
  files.
  
  
  
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[Asterisk-Users] Will Echo problems EVER be solved, I'm scared

[canuck15]

 
I came into this 
with my eyes wide open.  I have read ABSOLUTELY EVERYTHING there is to be 
found on the net about avoiding echo problems BEFORE I even attempted to create 
a production system.  Since lots of people are apparently using this in 
production environments now I just assumed that echo IS avoidable.  

 
As others have 
recommended, I created a test system with the proposed production parts.  I 
bought a couple different SIP phones to try and a Digium TDM01B card.  I am 
using an older PIII 1Ghz system with 815chipset (PCI Rev2.2) with 256MB for 
my test system.  The only thing that will be different on a production 
system is that I will be using a newer chipset PC with faster processor and 
512MB.  Probably Intel 7505, 7210, or 7211 chipsets which seem to be 
the most compatible with Asterisk.  
 
My problem is that I 
cannot eliminate echo no matter what I try.  I seriously doubt that a newer 
chipset faster PC with more memory will eliminate or even reduce my echo 
problems based on what I have read.   I am not about to drop more 
cash to try and find out.  Essentially, my findings are that Asterisk 
is NOT production capable for my configuration which is via FXO and PSTN.  
That is probably THE most common configuration so if it is not production 
capable like that it isn't production capable period as far as I'm 
concerned.  What a disappointment :(.  
 
Unless I am missing 
something I am sure that many many people with a similar configuration in a 
production environment have the same problem.  Perhaps they are just living 
with it??  For me it is just as unacceptable on an Asterisk system as it is 
on a traditional PBX.  Some calls are ok and some are not.  No 
correlation to local, long distance, time of day.  There always seems to be 
some echo.  Sometimes it is worse than other times.  Again, no 
correlation to local, long distance, time of day.  Tried connecting to ATA 
adapter and using VoIP provider instead to see if the telco was causing the 
problem.  That did not change anything.  Still the same general echo 
problem
 
The things I have 
tried include in no particular order and not limited to 
are:
 
*Buy latest TDM400P 
with latest FXO module
*Ensure copper 
connection to analog telco lines and telco are not causing problems including 
running a separate shielded line to the demarc AND having the telco guy come out 
and test the levels, impedance etc.
*Adjust RX/TX levels 
as per Asterisk Wiki using the quick Ztmonitor method and by using the detailed 
Ztmonitor method via a Telco 102milliwatt test phone #.  The end result was 
RX=8.0, TX=-1.0.  Since I still have echo problems I have tried all sort of 
other settings without success.
*After ALL of the 
above, try every possible combination of all of the following on Asterisk 
v1.0.9: echocancel (off, on, 128, 256, 16, 32, 64), echowhenbridged (on, off), 
echotraining (off, on, 800), Mark 2 (default, aggressive, CVS head 
developments, bugs.digium.com patches, adjust threshold level as per wiki etc. 
etc.)
*Make sure 
echotraining line is before FXO channel assignment in zapata.conf 
file
*Run fxotune which 
did not find a need to adjust the FXO levels 
(1=0,0,0,0,0,0,0,0)
 
Based on all the 
above testing the best settings were pretty much in line with what most people 
are finding. 
echocancel=on. 
echowhenbridged=on, echotraining=800, Mark 2 echo canceller, aggressive 
cancellation OFF, bugs.digium.com #2820 patch, RX=8.0, 
TX=-1.0.
 
Still have 
echo.  Aggressive mode helps a bit but then the other persons voice get's 
cut off a lot especially when I talk and the cutting in and out 
of the canceller is more noticeable and objectionable in general 
than if Aggressive is turned off.
 
I have two SIP 
phones.  An Aastra 9133i and a Grandstream GXP2000.  Echo problem is 
the same on both phones. 
 
 
I am located within 
a metropolitan area in Canada.
 
Any comments and/or suggestions would be greatly appreciated 
as I am pretty much out of ideas and ready to give up on Asterisk as a suitable 
traditional small business phone system replacement.
  

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[Asterisk-Users] AEL Question

[Greg Blakely]

I've been puttering around with extensions.ael, and had a 
question.  (Well, 2 questions, but they're related).
 
First, would asterisk recognize any other .ael 
files as asterisk extension language?
 
Second, is there a way to #include another file 
from extensions.ael like there is from extensions.conf?
 
TIA
 
 
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RE: [Asterisk-Users] AGI + Ruby

[Huddleston, Robert]

Actually Perl is even better
 
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Mark Phillips
> Sent: Wednesday, August 24, 2005 4:17 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] AGI + Ruby
> 
> Y'see it? There it goes! Right over his head.
> 
> 
> Huddleston, Robert wrote:
> > U joke - duh! 
> >  
> >  
> > 
> > 
> >>-Original Message-
> >>From: [EMAIL PROTECTED]
> >>[mailto:[EMAIL PROTECTED] On Behalf 
> Of Innocent 
> >>Evil
> >>Sent: Wednesday, August 24, 2005 3:53 PM
> >>To: Asterisk Users Mailing List - Non-Commercial Discussion
> >>Subject: Re: [Asterisk-Users] AGI + Ruby
> >>
> >>What IDE are you talking about?
> >>Any URL would be helpful.
> >>
> >>Thanks,
> >>
> >>
> >>
> >>
> >>>-Original Message-
> >>>From: [EMAIL PROTECTED]
> >>>Sent: Wed, 24 Aug 2005 15:16:18 -0400
> >>>To: asterisk-users@lists.digium.com
> >>>Subject: Re: [Asterisk-Users] AGI + Ruby
> >>>
> >>>I think you might find amethyst much simpler and possibly
> >>
> >>cheaper too. 
> >>
> >>>I believe the current IDE is 12.4K
> >>>
> >>>
> >>>
> >>>Innocent Evil wrote:
> >>>
> I would like to write AGI script in Ruby Would anybody
> >>
> >>please show
> >>
> me right direction..
> 
> 
> Thanks___
> Asterisk-Users mailing list
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>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> >>>
> >>>--
> >>>
> >>>Mark, G7LTT/KC2ENI
> >>>Randolph, NJ
> >>>http://www.g7ltt.com
> >>>___
> >>>Asterisk-Users mailing list
> >>>Asterisk-Users@lists.digium.com
> >>>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>>To UNSUBSCRIBE or update options visit:
> >>>   
> >>>
> >>
> >>http://lists.digium.com/mailman/listinfo/asterisk-users_
__
> >>
> >>>
> >>
> >>Asterisk-Users mailing list
> >>Asterisk-Users@lists.digium.com
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>To UNSUBSCRIBE or update options visit:
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> >>
> > 
> > ___
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> 
> -- 
> 
> Mark, G7LTT/KC2ENI
> Randolph, NJ
> http://www.g7ltt.com
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[Asterisk-Users] SIP Jitter Buffer on Asterisk

[Matt]

Am I correct in thinking that at this time the CVS-HEAD supports
Jitter Buffer for SIP on Asterisk?
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Re: [Asterisk-Users] AGI + Ruby

[Mark Phillips]

Y'see it? There it goes! Right over his head.


Huddleston, Robert wrote:
U joke - duh! 
 
 




-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
Innocent Evil

Sent: Wednesday, August 24, 2005 3:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AGI + Ruby

What IDE are you talking about?
Any URL would be helpful.

Thanks,





-Original Message-
From: [EMAIL PROTECTED]
Sent: Wed, 24 Aug 2005 15:16:18 -0400
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] AGI + Ruby

I think you might find amethyst much simpler and possibly 


cheaper too. 


I believe the current IDE is 12.4K



Innocent Evil wrote:

I would like to write AGI script in Ruby Would anybody 


please show 


me right direction..


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--

Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
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--

Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
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RE: [Asterisk-Users] AGI + Ruby

[Innocent Evil]

Well,, I never heard about 'amethyst'




> -Original Message-
> From: [EMAIL PROTECTED]
> Sent: Wed, 24 Aug 2005 15:59:12 -0400
> To: asterisk-users@lists.digium.com
> Subject: RE: [Asterisk-Users] AGI + Ruby
>
> U joke - duh!
>
>
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > Innocent Evil
> > Sent: Wednesday, August 24, 2005 3:53 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] AGI + Ruby
> >
> > What IDE are you talking about?
> > Any URL would be helpful.
> >
> > Thanks,
> >
> >
> >
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> > > Sent: Wed, 24 Aug 2005 15:16:18 -0400
> > > To: asterisk-users@lists.digium.com
> > > Subject: Re: [Asterisk-Users] AGI + Ruby
> > >
> > > I think you might find amethyst much simpler and possibly
> > cheaper too.
> > > I believe the current IDE is 12.4K
> > >
> > >
> > >
> > > Innocent Evil wrote:
> > > > I would like to write AGI script in Ruby Would anybody
> > please show
> > > > me right direction..
> > > >
> > > >
> > > > Thanks___
> > > > Asterisk-Users mailing list
> > > > Asterisk-Users@lists.digium.com
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > To UNSUBSCRIBE or update options visit:
> > > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > >
> > > --
> > >
> > > Mark, G7LTT/KC2ENI
> > > Randolph, NJ
> > > http://www.g7ltt.com
> > > ___
> > > Asterisk-Users mailing list
> > > Asterisk-Users@lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
> > >
> > >
> > http://lists.digium.com/mailman/listinfo/asterisk-users___
> > > 
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
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Re: [Asterisk-Users] chan_capi on slackware10? cannot compile :-( why?

[Goran Dj.]

> > capi6208   0
> > kernelcapi 30496   1  [capi]
> > capiutil   22272   0  [kernelcapi]
> > uhci   2   0  (unused)
> > usbcore59308   1  [uhci]
> > hisax 448240   0  (unused)
> > isdn  116684   0  [hisax]
> > slhc4976   0  [isdn]
> > wcfxo   8384   2
> > zaptel176992   8  [wcfxo]
> > ide-scsi9328   0
> > ne  6672   1
> > 83906000   0  [ne]
> > crc32   2880   0  [8390]
> > isa-pnp30736   0  [hisax ne]
> >
> > So, where is a problem? Should I compile kernel with capi as a part
of a
> > kernel, not as a module? How to do that?
>
> It's okay to use it as modules. But the cards supported by HiSax do
not
> provide CAPI interface. I don't know the status of mISDN, but that
would
> be the driver supporting CAPI.


Hmmm? I don't know what hisax doing here (and even what is that). My
ISDN card (winbond w6692cf chip) isn't in computer, I will put it there
when I successfully complile chan_capi. What modules do I need? Only
capi(&kernelcapi&caputil) and chan_capi?




>
> You don't have libcapi20 (or the development package of it) installed.


Yes, but I dont have it on my Slackware10 CD'es. I don't have libcapi,
or isdn4... or anything with isdn or capi in their name. Where to find
libcapi20 (od devel...) for slackware?

Goran


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Re: [Asterisk-Users] chan_capi on slackware10? cannot compile :-( why?

[Bob Goddard]

On Wednesday 24 Aug 2005 17:22, Goran Dj. wrote:
> I'm trying to compile chan_capi-0.5.4 on Slackware 10, but I have bunch
> of errors.
> (By the way, can I use chan_capi for ISDN card with winbond w6692cf
> chipset?)
>
> I'm not a linux expert, still :-)
> Before compiling, when I type "modprobe capi" to load capi module, and
> then "lsmod", i get list of modules:
[...]
> chan_capi.o chan_capi.c
> chan_capi.c:49:20: capi20.h: No such file or directory

The above is a hint.

You should have capi4linux installed.


B
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Re: [Asterisk-Users] RealTime ignoringswitch=> Realtime/[EMAIL PROTECTED] altime_ext

[Robert Webb]

On Wed, 24 Aug 2005 15:25:15 -0400
 John Novack <[EMAIL PROTECTED]> wrote:

In my case, mysql is set to "any" host

So, yes, it does seem to be an Asterisk issue

And my buddy is pretty savvy with mysql, Linux and 
databases on Unix/Linux, having worked for a large IT 
company for some 20 years.


John Novack
P.S. Robert- Something wrong with your mail clock?
You responded to a message hours before it was sent!




Sorry, was not trying to insult your expertise.. Just 
sometimes things can get overlooked.


Thanks for the heads up.. Something went awry with my 
email server today and the ntp client went screwy. Should 
be fixed now.


Robert
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Re: [Asterisk-Users] Warning Unable to allocate socket

[Bob Goddard]

On Wednesday 24 Aug 2005 13:40, Kamran Ahmad wrote:
> hello
>
> i m getting follwing messages in asterisk-1.0.9 after
> small interval. And i have to restart asterisk because
> after these errors asterisk cannot do any call. what
> is the reason calls are not going out. can u pls tel
> me how to solve this.

How many times are you going to ask this?

Someone has already replied to you.
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Re: [Asterisk-Users] DTMF not working

[Innocent Evil]

Hi Rob,

I am using RFC2833 everywhere including SIP phone, asterisk's sip.conf
Do you think, to raise the value from 100 to 400, would solve my issue?

Thanks,



> -Original Message-
> From: [EMAIL PROTECTED]
> Sent: Wed, 24 Aug 2005 08:46:43 -0700
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] DTMF not working
>
> Hi Mr. Evil,
>
> I'm not sure if the problem that I am describing relates to the problem
> that you are having.  It seems that when you press a key on a SIP phone
> that is set for inband DTMF, asterisk absorbs the tones until you
> release the key.  This way if you are using DTMF to do things like
> transfer calls, the user won't get tone blasts in their ear until
> asterisk has had a chance to interpret the tones.   After asterisk has
> figured out what to do with the tone, it generates and transmits it's
> own tones in the routine do_senddigit() (assuming that the DTMF tone
> should be passed on).  The duration of the DTMF tones that asterisk
> generates is fixed and independent of how long you pressed the key on
> your phone.
>
> In the line "!941+1336/100,!0/100", the 941 is one tone of the DTMF
> (dual tone multi-frequency), and 1336 is the other tone.  The 100 is the
> duration of those tones.   The tones are in Hz.  I'm not sure what units
> the duration is in, but I bumped mine from 100 to 400 and that seems to
> do the trick.  The part of the string that reads "!0/100" just shuts the
> tone generator off.
>
> Rob
>
> Innocent Evil wrote:
>
> >I am having same problem .. DTMF is not working from a SIP phone while
> >sending to Asterisk cmd VoiceMailMain.
> >
> >Would you please explain this line
> >"!941+1336/100,!0/100", /* 0 */
> >
> >what  value is what and how it affect on DTMF tone generation.
> >
> >Thanks,
> >
> >
> >
> >>I had a similar problem that seems to be caused by the DTMF tone
> lengths
> >>being to short.  Try this:
> >>
> >>Asterisk generates DTMF tones in  do_senddigit() in the file channel.c.
> >>The tones are defined in a const char array called dtmf_tones[].  Each
> >>DTMF tone is a string that looks something like:
> >>
> >>"!941+1336/100,!0/100", /* 0 */
> >>
> >>The part that reads !941+1336/100 is the part that you want.  Change
> the
> >>"100" to something bigger and recompile.  You will have to do that for
> >>every tone.   I'm using 400 right now, and it seems to be working.
> >>
> >>I hope that helps.
> >>
> >>Rob
> >>
> >>Peter Osborne wrote:
> >>
> >>
> >>
> >>>Hi all,
> >>>
> >>>I just upgraded from Asterisk 1.0RC1 to Asterisk 1.0.7 and our dtmf no
> >>>
> >>>
> >>longer
> >>
> >>
> >>>works with external phone systems. I have a Wildcard TDM400P with 4
> >>>
> >>>
> >>FXO's?
> >>
> >>
> >>>(it connects to analog lines). No changes were made to the config
> files.
> >>>
> >>>Here's my config:
> >>>
> >>>/etc/zaptel.conf
> >>>fxsks=1-4
> >>>loadzone = us
> >>>defaultzone=us
> >>>
> >>>/etc/asterisk/zapata.conf
> >>>[channels]
> >>>usecallerid=yes
> >>>hidecallerid=no
> >>>callwaiting=yes
> >>>usecallingpres=yes
> >>>threewaycalling=yes
> >>>transfer=yes
> >>>cancallforward=yes
> >>>callreturn=yes
> >>>echocancel=yes
> >>>echotraining=yes
> >>>rxgain=2.0
> >>>txgain=2.0
> >>>callgroup=1
> >>>pickupgroup=1
> >>>musiconhold=default
> >>>context=incoming
> >>>group=1
> >>>signalling=fxs_ks
> >>>echocancel=64
> >>>echocancelwhenbridged=yes
> >>>relaxdtmf=yes
> >>>channel => 1-3
> >>>
> >>>[pete_desk]
> >>>;Pete's Desk phone (Polycom IP 300)
> >>>type=friend
> >>>username=pete_desk
> >>>secret=pass
> >>>context=longdistance
> >>>callerid=Pete <601>
> >>>host=dynamic
> >>>mailbox=601
> >>>dtmfmode=inband
> >>>disallow=all
> >>>allow=ulaw
> >>>allow=alaw
> >>>
> >>>Thanks,
> >>>Pete
> >>>
> >>>
>
> --
> Robert Tarte
> Pacific CodeWorks
> P.O. Box 29050
> San Francisco, CA 94129
>
> (p) 831-426-7582
> (f) 831-426-7584
>
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RE: [Asterisk-Users] AGI + Ruby

[Huddleston, Robert]

U joke - duh! 
 
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Innocent Evil
> Sent: Wednesday, August 24, 2005 3:53 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] AGI + Ruby
> 
> What IDE are you talking about?
> Any URL would be helpful.
> 
> Thanks,
> 
> 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > Sent: Wed, 24 Aug 2005 15:16:18 -0400
> > To: asterisk-users@lists.digium.com
> > Subject: Re: [Asterisk-Users] AGI + Ruby
> >
> > I think you might find amethyst much simpler and possibly 
> cheaper too. 
> > I believe the current IDE is 12.4K
> >
> >
> >
> > Innocent Evil wrote:
> > > I would like to write AGI script in Ruby Would anybody 
> please show 
> > > me right direction..
> > >
> > >
> > > Thanks___
> > > Asterisk-Users mailing list
> > > Asterisk-Users@lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> > --
> >
> > Mark, G7LTT/KC2ENI
> > Randolph, NJ
> > http://www.g7ltt.com
> > ___
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >
> > 
> http://lists.digium.com/mailman/listinfo/asterisk-users___
> > 
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
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[Asterisk-Users] Music on hold configuration

[Anish Basu]

For some reason I cannot get mutiple moh classes to work.  My current
musiconhold.conf is:
[classes]
default => quietmp3:/var/lib/asterisk/mohmp3
company => quietmp3:/var/lib/asterisk/mohmp3/company

I am running Asterisk CVS-D2005.06.24.  When a caller is placed on hold, I
get the following message:
Aug 24 15:39:48 WARNING[21300]: res_musiconhold.c:870 local_ast_moh_start:
No class: company.

The class is correctly defined in musiconhold.conf.  The directory
/var/lib/asterisk/mohmp3/company exists with a single mp3 file in it.  The
dialplan contains the line: "exten => s, 1, SetMusicOnHold(company)".  I am
starting to run out of ideas.  Should I try using native format_mp3 instead
of mpg123?

Anish Basu
Field Systems Engineer
Softel, Inc.
Phone: (732) 705-9202
Cell: (732) 312-6634 

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Re: [Asterisk-Users] NAT and SIP.conf update.

[Bob Goddard]

On Wednesday 24 Aug 2005 09:44, razza wrote:
> I have a standard BT home DSL, which means I cannot have a static IP
> address, therefore i'm forced to use NAT, I subscribe to a DDNS service
> and have written a VB app which polls the router every 10 seconds and
> updates the DDNS if appropriate.

Ditch your ISP and go with one who will give you a static IP.


B
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Re: [Asterisk-Users] AGI + Ruby

[Innocent Evil]

What IDE are you talking about?
Any URL would be helpful.

Thanks,



> -Original Message-
> From: [EMAIL PROTECTED]
> Sent: Wed, 24 Aug 2005 15:16:18 -0400
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] AGI + Ruby
>
> I think you might find amethyst much simpler and possibly cheaper too. I
> believe the current IDE is 12.4K
>
>
>
> Innocent Evil wrote:
> > I would like to write AGI script in Ruby
> > Would anybody please show me right direction..
> >
> >
> > Thanks___
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> --
>
> Mark, G7LTT/KC2ENI
> Randolph, NJ
> http://www.g7ltt.com
> ___
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Re: [Asterisk-Users] HDLC/Zaptel/Kernel 2.6.11(.9)

[Kevin P. Fleming]

Matt Schulte wrote:

I see, any suggestions if he went up or down in version?


I could not make it work with 2.6.12.3/4/5, so I switched to 'raw' (no 
encapsulation) and it's worked beautifully since then.


This is not an option for you if you need to run protocols other than IP 
or if you need to connect to another device that demands Cisco 
encapsulation, but if you control both ends it's a better choice since 
it's more efficient as well.

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[Asterisk-Users] ISDN MSN problems

[Andrzej Nowrot]

Hi,

I`m new to all asterisk issues and I`m having this small problem :)
In my company I`m using ISDN phone and when I signed the fixed MSM (to a phone) 
I was not able to receive any calls. My dialplan is so defined that after 30s 
of ringing voicemail goes on, and when someone dial my ISDN (with MSM), goes 
direct to voicemail. I think is probably connected with issues in Zapata.conf 
(prefix, nationalprefix), but I`m not sure.

Could anyone help me with this problem?

Best regards 


--
INTERIA.PL. Portal z najlepsza wyszukiwarka. >>>
>>> http://link.interia.pl/f18ae

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Re: [Asterisk-Users] chan_capi on slackware10? cannot compile :-( why?

[Armin Schindler]

On Wed, 24 Aug 2005, Goran Dj. wrote:
> I'm trying to compile chan_capi-0.5.4 on Slackware 10, but I have bunch
> of errors.
>
> I'm not a linux expert, still :-)
> Before compiling, when I type "modprobe capi" to load capi module, and
> then "lsmod", i get list of modules:
> 
> capi6208   0
> kernelcapi 30496   1  [capi]
> capiutil   22272   0  [kernelcapi]
> uhci   2   0  (unused)
> usbcore59308   1  [uhci]
> hisax 448240   0  (unused)
> isdn  116684   0  [hisax]
> slhc4976   0  [isdn]
> wcfxo   8384   2
> zaptel176992   8  [wcfxo]
> ide-scsi9328   0
> ne  6672   1
> 83906000   0  [ne]
> crc32   2880   0  [8390]
> isa-pnp30736   0  [hisax ne]
> 
> So, where is a problem? Should I compile kernel with capi as a part of a
> kernel, not as a module? How to do that?

It's okay to use it as modules. But the cards supported by HiSax do not 
provide CAPI interface. I don't know the status of mISDN, but that would
be the driver supporting CAPI.
 
> Errors when I try to compile chan_capi:
> 
> [EMAIL PROTECTED]:#make
> ./create_config.sh "/usr/include"
> Checking Asterisk version...
>  * no 'struct ast_channel_tech', using old pvt
>  * ast_dsp_process() without 'needlock'
>  * no 'struct ast_callerid'
>  * found 'struct timeval delivery'
>  * no 'transfercapability'
>  * no 'ast_config_load'
> config.h complete.
> gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g  -I/usr/i
> nclude -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586  -DASTERISKVERSION=\"\
> " -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO   -c -o
> chan_capi.o chan_capi.c
> chan_capi.c:49:20: capi20.h: No such file or directory

You don't have libcapi20 (or the development package of it) installed.

Armin

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Re: [Asterisk-Users] RealTime ignoringswitch=> Realtime/[EMAIL PROTECTED] altime_ext

[John Novack]

In my case, mysql is set to
"any" host

So, yes, it does seem to be an Asterisk issue

And my buddy is pretty savvy with mysql, Linux and databases on
Unix/Linux, having worked for a large IT company for some 20 years.

John Novack
P.S. Robert- Something wrong with your mail clock?
You responded to a message hours before it was sent!


Robert Webb wrote:

On Wed, 24 Aug 2005 14:47:25 -0400
  
 "Araba, Michael" <[EMAIL PROTECTED]> wrote:
  
  Thanks John, You are my savior. This is such
a great relief. Apparently

realtime will not use either '127.0.0.1' or 'localhost' to connect to
the

database. I had to use the actual IP address attached to the NIC before
it

worked. 
  
  

  
  
You claim it is an Asterisk issue, did you by any chance make sure that
database was allowing connections on 127.0.0.1 and localhost and not
just the actual IP??
  
  
Robert
  
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Re: [Asterisk-Users] AGI + Ruby

[Mark Phillips]

I think you might find amethyst much simpler and possibly cheaper too. I 
believe the current IDE is 12.4K




Innocent Evil wrote:

I would like to write AGI script in Ruby
Would anybody please show me right direction..


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--

Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
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Re: [Asterisk-Users] RealTime ignoringswitch=>Realtime/[EMAIL PROTECTED] altime_ext

[John Novack]

Two of us swam in that pool for some hours.
Trying every possible combo until we  hit on the correct one.
I was simply trying to get the CDR to work with mysql, and my buddy 
wanted that as well as realtime. It isn't JUST realtime, but something 
in Asterisk. He has HEAD from late January, I from late February, and 
yours is pretty current.


Perhaps someone smarter than us can add that information to the Wiki, or 
report it as a bug, or SOMETHING?


John Novack

Araba, Michael wrote:


Thanks John, You are my savior. This is such a great relief. Apparently
realtime will not use either '127.0.0.1' or 'localhost' to connect to the
database. I had to use the actual IP address attached to the NIC before it
worked. 


My OS is Debian just a note and Asterisk HEAD from August 20, 2005

Details below for those who might be swimming in the same pool with me.

res_mysql.conf settings
[general]
dbhost = 
dbname = RealTimeMaster
dbuser = xxx
dbpass = xxx
dbport = 3306
dbsock = /tmp/mysql.sock

extconf.conf
[settings]
voicemail => mysql,RealTimeMaster,voicemail
sipusers => mysql,RealTimeMaster,sip_buddies
sippeers => mysql,RealTimeMaster,sip_buddies
queues => mysql,RealTimeMaster,queue
queue_members => mysql,RealTimeMaster,queue_member

cdr_mysql.conf
[global]
hostname=localhost
dbname=RealTimeMaster
table=cdr
password=x
user=xx
;port=3306
;sock=/tmp/mysql.sock
;userfield=1


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Re: [Asterisk-Users] RealTime ignoringswitch=> Realtime/[EMAIL PROTECTED] altime_ext

[Robert Webb]

On Wed, 24 Aug 2005 14:47:25 -0400
 "Araba, Michael" <[EMAIL PROTECTED]> wrote:
Thanks John, You are my savior. This is such a great 
relief. Apparently
realtime will not use either '127.0.0.1' or 'localhost' 
to connect to the
database. I had to use the actual IP address attached to 
the NIC before it
worked. 





You claim it is an Asterisk issue, did you by any chance 
make sure that database was allowing connections on 
127.0.0.1 and localhost and not just the actual IP??


Robert
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RE: [Asterisk-Users] Re: [Serusers] SER IP PBX for multiple clients

[Ronald Voermans]

Waldo,

How do you let your customers manage 'their' PBX. I too have a setup
like you. However, I installed a * server for each customer, via
vserver. I'd like to now what kind of software/webbased package you use
for this.

I also have SER installed as a front-end server for the * servers. But,
as I'm still not very into SER, don't know exactly how this fits in.
Should I use SER only as proxy, or also as a registrar server (with the
same problems as you describe)?

Hope someone at this list is able to help us! 

Regards,
Ronald

-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Waldo Rubinstein
Verzonden: woensdag 24 augustus 2005 17:28
Aan: Iqbal
CC: SER User Mailing List Mailing List; Asterisk Users Mailing List -
Non-Commercial Discussion
Onderwerp: [Asterisk-Users] Re: [Serusers] SER IP PBX for multiple
clients

lqbal,

I do plan on having alot of users. Two markets I'm trying to get some
volume users from are: residential consumers and business users.  
Residential consumers should get basic line services such as their own
DID, voicemail, caller-id, call-waiting, three-way calling, and
basically, all the standard features you get from companies like Vonage,
etc. This particular market base will have a higher volume than business
users.

Business users will get everything residential consumers get, plus
additional features. Features such as, automated attendant, extension-
to-extension calling, company directories, etc.

I guess I would need to have SER and Asterisk work in tandem. Now, what
should be the correct approach in assigning responsibilities to both SER
and Asterisk respectively? Should SER be used strictly as proxy to
Asterisk, may be also registrar, and NAT helper, and then have Asterisk
handle all the calling plans, features, enhanced services and SER will
simply forward everything to Asterisk? Can you or someone advise as to
what would be the more robust/scaleable architecture to deploy this?
Needless to say, it is imperative that I get proper CDR from either one
or both systems in order for me to properly bill our users. I don't know
which of the two platforms has a more robust/customizable call logging
facility.

I took the liberty of cross-posting to the Asterisk list in order to get
some of their feedback as well.

Thanks,
Waldo

On Aug 23, 2005, at 6:49 AM, Iqbal wrote:

> Um..no actually I am saying you could combine both, but that will only

> help if you have alot of users. I guess you could direct calls to a 
> particular sip client, ut normally when ser and asterisk work in 
> tandem, all calls from SER hit one section of sip.conf, and hence can 
> only be pointed to one context, you can get around this by including 
> contexts from this default one, which is what I do, based upon a mysql

> lookup, but then you will have problems in call pickup, because all 
> pickup is not context based, again there is a solution to this, if you

> look at bristuff patch for asterisk.
>
> If you dont have many users stick with ust asterisk, if you want to 
> scale you may need to kludge something with ser and asterisk, and this

> might be easy or hard depending on exacly what you require, and call 
> scenarios.
>
> Iqbal
>
> Waldo Rubinstein wrote:
>
>
>> The way I manage this in Asterisk is every SIP UA has a unique login

>> but in different contexts. I suppose that if SER directs a call to  
>> Asterisk to the specific SIP client, Asterisk will recognize it  
>> belongs to a different context. The question is, I don't know if SER

>> knows about multiple contexts under the premise of the Asterisk 
>> world.
>>
>> Also, I get the feeling you are pretty much telling me to stick to  
>> Asterisk :) Is that so?
>>
>> Thanks,
>> Waldo
>>
>> On Aug 22, 2005, at 3:26 PM, Iqbal wrote:
>>
>>
>>> Hi
>>>
>>> If you are already using multiple contexts within asterisk, then   
>>> your already half way there, the problem is if you stick in SER,   
>>> bcause then  your phones are not registered in asterisk, hence all  
>>> fall into the same context in sip.conf, which means they all  will  
>>> hit  one context in extensions.conf, hence you should look into 
>>> that.
>>>
>>> I am not sure if you can do the 101/102 extension thing in   
>>> asterisk, since aliases will be bound to a contact, whereas in   
>>> asterisk the context is also part of the dialing plan.
>>>
>>> DID can be done, as can forking and directing to voicemail on no   
>>> answer.
>>>
>>> Iqbal
>>>
>>> Waldo Rubinstein wrote:
>>>
>>>
>>>
 Hello,

 I'm still trying to learn more about SER. I've been using Asterisk

 to  manage virtual PBX services for different companies by using  
 multiple  contexts within Asterisk. However, since I only use  
 Asterisk with SIP  UAs and to communicate with ITSPs, I don't have

 the need to have all  the fancy features Asterisk offers, plus I  
 have the additional  advantage of having the built-in NAT support  
 in SER.
>>>

[Asterisk-Users] zapata.conf for a BT phone line with a TDM422P

[Steve Dobson]

Hi guys

Forgive the newbie question but I am setting up my first * system :-)

I live in the UK with two standard BT phone lines.  The hardware used
is a TDM400P with two FXO and two FXS daughter boards.

Currently only one FXO and one FXS line is used during testing.

Incoming calls work fine, as do the phones plugged into the system.

However outbound calls do not work.  The number is dialled, and some
attempt is made to send the number, but I get the old BT "the number
you have dialled as not been recognised" message.

I had this problem years ago with my old Psion 3 - it could be used
to tone dial numbers using its speaker and it took me a while get it
to work by slowing the dialling down.  I'm hoping that the same is
true now.

Can someone explain what the different timing settings are.  What,
is wink and flash?  Is debounce the time between each tones?

Of course if someone has a working zapata.conf for TDM400P on a BT
land line then that would be really, really useful :-)

Ta in advance
Steve

-- 
Sex dumps core
(Sex is a Simple editor for X11)
-- Seen on debian bugtracking


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RE: [Asterisk-Users] RealTime ignoringswitch=>Realtime/[EMAIL PROTECTED] altime_ext

[Araba, Michael]

Thanks John, You are my savior. This is such a great relief. Apparently
realtime will not use either '127.0.0.1' or 'localhost' to connect to the
database. I had to use the actual IP address attached to the NIC before it
worked. 

My OS is Debian just a note and Asterisk HEAD from August 20, 2005

Details below for those who might be swimming in the same pool with me.

res_mysql.conf settings
[general]
dbhost = 
dbname = RealTimeMaster
dbuser = xxx
dbpass = xxx
dbport = 3306
dbsock = /tmp/mysql.sock

extconf.conf
[settings]
voicemail => mysql,RealTimeMaster,voicemail
sipusers => mysql,RealTimeMaster,sip_buddies
sippeers => mysql,RealTimeMaster,sip_buddies
queues => mysql,RealTimeMaster,queue
queue_members => mysql,RealTimeMaster,queue_member

cdr_mysql.conf
[global]
hostname=localhost
dbname=RealTimeMaster
table=cdr
password=x
user=xx
;port=3306
;sock=/tmp/mysql.sock
;userfield=1


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Re: [Asterisk-Users] Lots of console; attach and grep?

[Eric Wieling aka ManxPower]

Matthew Boehm wrote:

Eric Wieling aka ManxPower wrote:

You mean like the info in /var/log/asterisk which is configured via 
/etc/asterisk/logger.conf ?



Damn. If I change any logging, that's going to require an asterisk 
restart isn't it?


voip-1*CLI> logger reload
  == Parsing '/etc/asterisk/logger.conf': Not found (No such file or 
directory)

Asterisk Event Logger restarted
voip-1*CLI>


--
Eric Wieling * BTEL Consulting * 504-210-3699 x2120

r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care and don't insert
this by default into all your dial statements as you are killing call
progress information for the user. Really, you almost certainly do not
want to use this. Asterisk will generate ring tones automatically where
it is appropriate to do so. "r" makes it go the next step and
additionally generate ring tones where it is probably not appropriate to
do so.

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Re: [Asterisk-Users] SIP powercycle not hanging up

[Sharon]

Steve,
I have been experiencing the same issue. I tried the qualify option as
well but didn't help.
Let me know if you find any solution.
-AG


On 8/23/05, Steve Edwards <[EMAIL PROTECTED]> wrote:
> I have Sipura 841's talking to a CVS-HEAD of August 4.
> 
> If I disconnect the power to the Sipura, Asterisk does not hang up the
> channel.
> 
> My sip.conf for this phone looks like:
> 
> ;
> [super1]; Sipura 841
>   disallow   = all
>  allow   = ulaw
>  callerid= "super1"
>  context = inside
>  dtmfmode= rfc2833
>  host= dynamic
>  qualify = 200
>  secret  = xx
>  type= friend
>  username= super1
> 
> I added the "qualify" but that did not help.
> 
> Any clues?
> 
> Thanks in advance,
> 
> Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
> Newline   [EMAIL PROTECTED]Fax: +1-760-731-3000
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[Asterisk-Users] saynumber and variables

[Daniel]

Hello
Is there any way to associate a txt file to a varible to run Saynumber
command?



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Re: [Asterisk-Users] Sql Realtime

[Matthew Boehm]

Jimmy Smith wrote:


.. options i found would be to use mysql  from the asterisk distro..
but are the memory leaks fixed ?


Options are to use res_config_mysql found in asterisk-addons.

What memory leaks?

-Matthew

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Re: [Asterisk-Users] Lots of console; attach and grep?

[Matthew Boehm]

Eric Wieling aka ManxPower wrote:

You mean like the info in /var/log/asterisk which is configured via 
/etc/asterisk/logger.conf ?


Damn. If I change any logging, that's going to require an asterisk 
restart isn't it?


-Matthew

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[Asterisk-Users] c++ class for agi?

[Paul Zimm]

Being a lazy person, I was wondering if anyone has a c++ class for 
interfacing with asterisk AGI? I am aware of the C library listed on the 
wiki.

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RE: [Asterisk-Users] Cisco 7960 / SIP & tftp configs

[Tarpo, Louie]

I've not had the deleting lines problem.  When you're deleting a line, are you 
changing the config to ... 
line6_name: ""
line6_displayname: ""
line6_shortname: ""
line6_authname: ""
line6_password: ""


#Change lineX_shortname: "" to whatever you want them to see on the LCD.
line4_name: ""
line4_displayname: ""
line4_shortname: "Line4"
line4_authname: ""
line4_password: "password"



#Change  to your VoiceMailMain() extension
messages_uri: 


Louie


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Asterisk
User Group
Sent: Wednesday, August 24, 2005 10:45 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Cisco 7960 / SIP & tftp configs


I have three questions about my 7960 phone that I can't discern from the 
docs/wiki.

1st - If I change the SIPxx.cnf file to change registrations it sets 
up new lines as expected. If I delete a line it doesn't get removed when 
I reboot the phone. I have to go to the phone, unlock it, and reset the 
SIP parameters. How do I make it "forget" what it has programmed and 
listen only to the download?

2nd - Has anyone figured out how to get the Message button to launch a 
dial to VoicemailMain?

3rd - How do I display on the LCD an alias to the registered line?
line1_name: 2000
line1_authname: "2000"
line1_password: **

The doc seems to suggest that line1_name is what it registers with and 
line1_authname is what it uses "if challenged during the 
authentication". This doesn't make any sense to me. I am looking for the 
line to be "2000" but the display to say "Home" or "Business", etc.

Thanks, dbc.
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Re: [Asterisk-Users] Google introduces text/audio chat client andservice

[Douglas Logan]

Couldn't you already integrate Asterisk and GoogleTalk with this?
http://www.jivesoftware.org/index.jsp since GoogleTalk just uses XMPP?

On 8/24/05, Andreas Bayer <[EMAIL PROTECTED]> wrote:
> Am Mittwoch, 24. August 2005 10:32 schrieb Mat Stace, Colewood Internet:
> > Lifted from the developer page of the google talk site
> > (http://www.google.com/talk/developer.html)
> >
> > 5. What protocols are used for voice calls?
> >
> > Google Talk supports a custom XMPP-based signaling protocol and
> > peer-to-peer communication mechanism. We will fully document this protocol.
> > In the near future, we plan to support SIP signaling.
> >
> i think the xmpp signaling protocol is the "standard" open protocol for voice
> messages with jabber.
> apples ichat use the same protocol. ichat can use it for audio and video
> chats.
> 
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RE: [Asterisk-Users] Cisco 7960 / SIP & tftp configs

[Matt Schulte]

1) You have to do a factory reset, or wipe out the line config. 

2) By default it dials ext 8500 I believe.

3) You *should* be able to change _name, I can't remember the effect
that has since you already have authname in.

Matt 

-Original Message-
From: Asterisk User Group [mailto:[EMAIL PROTECTED] 
Sent: Wednesday, August 24, 2005 11:45 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Cisco 7960 / SIP & tftp configs

I have three questions about my 7960 phone that I can't discern from the
docs/wiki.

1st - If I change the SIPxx.cnf file to change registrations it sets
up new lines as expected. If I delete a line it doesn't get removed when
I reboot the phone. I have to go to the phone, unlock it, and reset the
SIP parameters. How do I make it "forget" what it has programmed and
listen only to the download?

2nd - Has anyone figured out how to get the Message button to launch a
dial to VoicemailMain?

3rd - How do I display on the LCD an alias to the registered line?
line1_name: 2000
line1_authname: "2000"
line1_password: **

The doc seems to suggest that line1_name is what it registers with and
line1_authname is what it uses "if challenged during the
authentication". This doesn't make any sense to me. I am looking for the
line to be "2000" but the display to say "Home" or "Business", etc.

Thanks, dbc.
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RE: [Asterisk-Users] Yellow Alarm issues with second TE410P installed

[Asterisk]

oops - I meant to says "Channels 25 - 48" are going into yellow alarm

Bart

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Asterisk
Sent: Wednesday, August 24, 2005 10:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Yellow Alarm issues with second TE410P
installed


Hi,

Ok, I put the new TE410 into our system last night - This is the second
TE410P card.  There is also TDM400P in Slot 5 with IRQ 9 which is shared
with aic7xxx (SCSI)

According to "cat /proc/interrupts", the new card is in the second slot
using IRQ 11 (the first card is in slot 1 on IRQ 10  - but always has been).

With only the original card install, there has been no issues with alarms.
Currently, I have card 2 Looped back to each other - no active connections -
Port 1 => 3 & Port 2 => 4)

I came in this morning and channel 25 has been going into Yellow Alarm on
and off since midnight according to the logs. (12 times from 4:13 am to 8:04
Pacific)

Card 1 has Telco provided T1's on Port 3 & 4  and in-house phone system on
Port 1 and 2.  Asterisk is program to route call from Port 3 or 4 to the
appropriate in-house system according to the DNIS digits received.

All T1's are setup as D4, AMI, SF and E&M Wink.  The timing source is Port 4
on Card 1 - but not sure if this is needed.

Any clues what I should look at now?

Thanks

Bart



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Re: [Asterisk-Users] Lots of console; attach and grep?

[Jimmy Smith]

hehe yes exactly..

you could tail -f that file .. or grep
as in 

tail -f /var/log/asterisk/verbose |grep -10 -v 'somestring'

that would give you 10 lines around it.. or before it i dont remmeber
off the bat..


On 8/24/05, Eric Wieling aka ManxPower <[EMAIL PROTECTED]> wrote:
> Matthew Boehm wrote:
> > We have recently started routing about 3 PRI's worth of traffic thru our
> > asterisk box.
> >
> > The text on the console now flys by so damn fast, I can't really see
> > what the heck is going on. Even with verbosity 0 and debug 0 it is still
> > so fast.
> >
> > Is there some way I can attach to the console in a way that will allow
> > me to grep or otherwise filter the text so I can focus on something in
> > particular?
> 
> You mean like the info in /var/log/asterisk which is configured via
> /etc/asterisk/logger.conf ?
> 
> 
> --
> Eric Wieling * BTEL Consulting * 504-210-3699 x2120
> 
> r: Generate a ringing tone for the calling party, passing no audio from
> the called channel(s) until one answers. Use with care and don't insert
> this by default into all your dial statements as you are killing call
> progress information for the user. Really, you almost certainly do not
> want to use this. Asterisk will generate ring tones automatically where
> it is appropriate to do so. "r" makes it go the next step and
> additionally generate ring tones where it is probably not appropriate to
> do so.
> 
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Re: [Asterisk-Users] Cisco 7960 / SIP & tftp configs

[Asterisk]

I'm not in the office at the moment to make sure, but if memory serves,

to set a value to 'nothing or null'
line1_name: "UNPROVISIONED"

messages_uri: 123
where 123 is in extensions.conf as 
exten => 123,1,VoiceMailMain(${CALLERIDNUM})
or something similar

line1_shortname: "Alias"


Best Regards,
Ben



- Original Message -
From: Asterisk User Group
To: 
Subject: [Asterisk-Users] Cisco 7960 / SIP & tftp configs
Sent: 8/24/2005 1:05:59 PM

I have three questions about my 7960 phone that I can't discern from the 
docs/wiki.

1st - If I change the SIPxx.cnf file to change registrations it sets 
up new lines as expected. If I delete a line it doesn't get removed when 
I reboot the phone. I have to go to the phone, unlock it, and reset the 
SIP parameters. How do I make it "forget" what it has programmed and 
listen only to the download?

2nd - Has anyone figured out how to get the Message button to launch a 
dial to VoicemailMain?

3rd - How do I display on the LCD an alias to the registered line?
line1_name: 2000
line1_authname: "2000"
line1_password: **

The doc seems to suggest that line1_name is what it registers with and 
line1_authname is what it uses "if challenged during the 
authentication". This doesn't make any sense to me. I am looking for the 
line to be "2000" but the display to say "Home" or "Business", etc.

Thanks, dbc.
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