date:20060401

Rudolf Ladyzhenskii wrote:
> I am not. I have one license and use i channel.
> It seems to detect the fact there are no more channels left and keeps
> warning me about it in case I want to use more.

I reviewed the code for that module after reading your original message,
and confirmed that it will only generate that message if you try to
create and use more g729 transcoder channels than you have licenses for.

Do you have monitoring or anything else running on the channel that
might cause another transcoder path to be needed?
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[Asterisk-Users] G729 Passthrough question

2006-04-01 Thread From PH

hi group,

is there a way that SIP phones be allowed to use G.729 passthrough when
calling each other and when calling PSTN through Zap that asterisk
force the phones to use ulaw.

thanks,

ultor
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re: [Asterisk-Users] Re: no audio

2006-04-01 Thread Alyed Tzompa


		That was a bug fixed in Asterisk version 1.2.3  recently version 1.2.6 was released, so don't worry you can try the latest one without timing fears :DAlyed 
		
		

Return-Path: <[EMAIL PROTECTED]> Sat Apr 01 15:42:39 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP;Sat, 1 Apr 2006 15:42:39 -0700
		Wierd timing - I'm struggling with exactly the same issue. My problemwas with ZAP - ZAP. The phones ring, but no audio. Turns out there'sa bug with the version I'm running. It has to do w/ the system date. When I changed my system date to 1-Jan-06, everything worked!! Here'swhat I found from another posting:>this is a new bug which is submitted: http://bugs.digium.com/view.php?id=6349>change your system date to an older value. everything will work again.I'm hoping the bug is fixed in a more recent release build, but Ihaven't tried yet.Yours,HughOn 4/1/06, Luis herrera <[EMAIL PROTECTED]>wrote:> Hi. I have a [EMAIL PROTECTED] setup at my home. My problems is with> phones outside my network. I call the extensions> without a problem, it rings but when they answer I> can't hear them and they can hear me.> I set up in the SIP.CONF> nat=yes>> I'm I missing any other setting or do I need a special> switch that support asterisk.> Thank you for your help.>>> __> Do You Yahoo!?> Tired of spam? Yahoo! Mail has the best spam protection around> http://mail.yahoo.com> ___> --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:> http://lists.digium.com/mailman/listinfo/asterisk-users>___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] G729 codec problems

2006-04-01 Thread Alyed Tzompa

I used g729 couple of times in the past and got the warning messages ONLY when I was trying to use more channels than the total amount of licenses I'd got.If you are sure you are using only one device that needs the license, I would suggest to check out how it is communicating with Asterisk. Also, if you have enough time try using the g729 with another soft / IP phone and see if you get the same result.Alyed ---I am not. I have one license and use i channel.It seems to detect the fact there are no more channels left and keepswarning me about it in case I want to use more.It is fine, but the warning is constant. All you see on Asteriskconsole is running warning message.RudolfOn 4/2/06, Kevin P. Fleming <[EMAIL PROTECTED]>wrote:> RumaTech wrote:>> > And it keeps running like that. Call usually come through OK. If i try> > to use "show g729" command, it shows that all codecs are in use. Well,> > this is fine, I am using one, but I do not want to see those warnings.> > Once is quite enough. Those continuos warnings make it impossible to se> > any other asterisk output. How does one turns them off?>> You can't make them stop except by not trying to use more channels than> you have licenses for.> ___> --Bandwidth and Colocation provided by Easynews.com -->> Asterisk-Users mailing list> To UNSUBSCRIBE or update options visit:> http://lists.digium.com/mailman/listinfo/asterisk-users>___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] Problem: ringtones stop unexpectedly

2006-04-01 Thread Alyed Tzompa


		Have you tryed phoning a fixed line instead of a cell phone?is this giving the same result?I assume your outgoing call to a the cellphone goes through a Zap channel. Try another one (e.g. Zap channel 2), and let us know the result.Alyed 
		
		

Return-Path: <[EMAIL PROTECTED]> Sat Apr 01 18:47:36 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP;Sat, 1 Apr 2006 18:47:36 -0700
		
		I should've mentioned that before. I've tried doing that and it has noeffect. I've tried both upper and lower-case 'r's.I've also tried a workaround that I thought would work, but it doesn't:Answering the call and then using the playtones(ringing) command beforeconnecting to my cellphone. -Original Message-Date: Sat, 1 Apr 2006 19:59:46 +0100From: "Julian J. M." <[EMAIL PROTECTED]>Subject: Re: [Asterisk-Users] Problem: ringtones stop unexpectedlywhen multiple channels are dialedTo: "Asterisk Users Mailing List - Non-Commercial Discussion"Message-ID:<[EMAIL PROTECTED]>Content-Type: text/plain; charset=ISO-8859-1Try adding 'r' to the dial options. According to "show application dial":r - Indicate ringing to the calling party. Pass no audio to thecallingparty until the called channel has answered.exten => 3058472194,1,Dial(SIP/1035&SIP/[EMAIL PROTECTED],50, r)Julian.On 4/1/06, Carlos A. Alfaro <[EMAIL PROTECTED]>wrote: Hello Everyone. I usually find my own solutions for problems but thistime,> after several months, I've given up. My asterisk is set up so that incoming calls from my voip provider ring on> both my sip extension and my cellphone at the same time. When the system> receives an incoming call, ringtones indicating that the call is being> connected play normally for the first 5 seconds to the caller, but they> suddenly stop as the call to my cellphone starts to make progress. This> causes some people to hang up, despite the fact that the call is stillbeing> connected. Callers who stay on the line are able to talk to me on either> the sip extension or the cellphone once I pick up either one. I have tried a lot of workarounds like including a priority to answer the> incoming call, invoke the playtones command before the dial command, but> this doesn't seem to work either. Can anyone replicate the problem? HaveI> ran into a bug? I have pasted as much info as I deemed relevant; pleaselet> me know if I'm missing something. Thanks.--___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-usersEnd of Asterisk-Users Digest, Vol 21, Issue 2*___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] Install problem with res_snmp.so from current trunk (bug?)




Kevin P. Fleming wrote:

Rich Adamson wrote:


Is this worthy of opening a bug assuming the above comment is still
valid?  Would the individual(s) maintaining res_snmp want to log into
either of these internet accessible boxes to identify the root cause?


The module loader in trunk is undergoing changes that will eliminate
this problem very soon.


No problem. Thanks.

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RE: [Asterisk-Users] *.conf generator

2006-04-01 Thread mustardman29

Because Aslinux is an embedded solution just like most PBX's. 

> -Original Message-
> From: Matt [mailto:[EMAIL PROTECTED] 
> Sent: Saturday, April 01, 2006 12:53 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] *.conf generator
> 
> Other thing I'm thinking... why are you running astlinux?  
> Asterisk really isn't that hard to install..
> 
> make; make install; make config
> 
> 
> On 4/1/06, Matt <[EMAIL PROTECTED]> wrote:
> > I don't understand your question.   You don't want to generate the
> > config files by hand, but yet you can't use FreePBX?   Why 
> can FreePBX
> > not generate the conf files and then you go on to use astlinux?
> > FreePBX should run on any linux distribution.
> >
> >
> > On 4/1/06, mustardman29 <[EMAIL PROTECTED]> wrote:
> > >
> > > Is there a good free *.conf generator out there.  Manual 
> > > configuration is just too tedious.  I run Astlinux so a 
> lot of the 
> > > GUI's such as AMP (FreePBX)are not an option either.
> > >
> > > I used to use IPManager which did a great job but that 
> project has 
> > > been discontinued :(.
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> > > Asterisk-Users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> 
> 
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Re: [Asterisk-Users] voicemail to email sending problems

2006-04-01 Thread C F

What does your /var/spool/mqueue look like? The messages in there
should give you a clue what's wrong. For some reasone I'm suspecting
that your exchange server is behind the same NAT as the sendmail
machine, sendmail is resloving yourdomain.com to the public MX record,
and your NAT device (like most NAT servers) will not allow internal
access to the public ip address that is mapped internaly. The best way
to work around *all* of these problems is to:
1. Configure the sendmail.cf (using of course the *.mc files,but if
you want to edit the .cf file it's macro DS change that to
read:DSipaddress) to use a smart host that will allow relay from this
sendmail box. In your case create a rule on your exchange email to
allow relaying from your sendmail box, and configure your sendmail box
to use the exchange server using the IP address of the exchange
server.
2. Make sure that the sendmail box uses an email address that is *not*
local to the sendmail box, and is real. If you run exchange it is
usualy the easiest to do this by configuring masquardating on the
sendmail box (again use the .mc files, but if you edit the .cf I think
it's the DM option), and create a secondary email address for someone
with root at yourdomain.com. That way all the NDRs will always go
there.

On 4/1/06, Jordan Novak <[EMAIL PROTECTED]> wrote:
>
>
>
> I have a box that will send to my personal pop/web based email but will not
> send to my exchange server. I have checked the MX record and DNS settings. I
> know there is something you can do like this to check it but it returns
> either a -1 or 0 (have no idea what that means)
>
>  sendmail
>  /mx
>
>  anyway I can send to a ISP based Mail account outside the company. We have
> .wav files allowed we also require smtp authentication. We do have an IP
> that is allowed to accept non authenticated mail from our databases, but I
> am not sure how to use this address with sendamil instaed of it using the MX
> record. which is mail.timbucktoo.com instaed of the allowed ip to bounce off
> of.
>  Should I be formatting the address in voicemail.conf to the allowed IP?
>  so its formatted [EMAIL PROTECTED]
>  Any thoughts.
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Re: [Asterisk-Users] Panasonic KXTD 1232 6

2006-04-01 Thread C F

FYI, the Panasonic 1232 system is discontinued.

On 4/1/06, Krzysztof Drewicz <[EMAIL PROTECTED]> wrote:
> charles napisaĹ�(a):
>
> > I want to replace a Telebutler software
>
> It's Telebutler software some simple IVR/CC solution?
>
> > auto attendent system that used a 4 port Dialogic board connected to a
> > Panasonic KXTD 1232 6 line system. We have spare computers here. How
> > do I connect asterisk to this Panasonic system?
>
>
> IIRC KXTD 1232 is a 12 public, 32 inside access PBX (up to 24 system and
> 8 pots for faxes or sth like that).
> It uses 6 BRI to access PTSN lines so you have to use some multi-port
> BRI (active and powered) card.
> http://www.eicon.com/worldwide/products/MediaGateways/diva-server-v4bri.htm
> or
> http://www.beronet.com/index.php?option=com_content&task=view&id=40&Itemid=28&lang=en
> The latter costs:
> BN8S0 8 Port S0 Card (TE/NT) + - Power Bundles BN Power
> 984,84EUR [incl. VAT] 849,00EUR [excl.VAT]
>
> This is not very cost-effective solution, and in most cases you have to
> use chan_misdn or other not-so-very-popular channel driver.
>
> Why you use plain PBX when you could do as simple as: one two/four span
> E1/T1 card, one port connected to Telco, second (3rd,4rd) connected to
> channel bank ?
>
>
> Btw: you could have a 4 times E1 in one PCI card (brand new) for less
> than a 650 USD and some Zhone CB from ebay around 120-150 USD (not very
> much used, i'm using it by my self).
>
>
> --
> Krzysztof Drewicz
>
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Re: [Asterisk-Users] H323 on way voice

2006-04-01 Thread Jeremy McNamara


Il Neofita wrote:


Hi,
I installed H323, however when I make a call from SIP Phone -> 
Asterisk H323 -> Provider H323 the provider can hear me, but I cannot 
hear nothing.
The asterisk is 1.2.6 with G729 license, and the asterisk is connect 
direct to internet with a public IP.

Any thoughts?




Set a valid bindaddr
Ensure G.729 is actually getting allowed


If you expect any more assistance, at all, debug information is required 
- So for now I am totally guessing.




Jeremy McNamara
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Re: [Asterisk-Users] Zap channels - help

2006-04-01 Thread Tzafrir Cohen

On Fri, Mar 31, 2006 at 10:36:02PM -0300, Josué Conti wrote:
> I am installing one asterisk, to establish connection with my PABX Siemens,
> in ISDN, link went up normally, also I obtain to internally call the
> branches the PABX, normally, but when I try to dial for the PSTN, through
> pabx with the command  exten = _ 19, 1,
> dial(zap/g2/${EXTEN}, 30) asterisk, reports me the following error:
> -- Executing Dial("SIP/8110-a729", "zap/g2/1971411234|30") in new stack
> -- Called g2/1971411234
> -- Channel 0/1, span 2 got hangup
> -- Hungup 'Zap/32-1'
>   == No one is available to answer at this
> time
>  However, when use the rule exten = _ 7xxx, 1, dial(zap/g2/${EXTEN}, 30) I
> obtain to call the branches pabx, normally.
> -- Executing Dial("SIP/8110-71ee", "zap/g2/7500|30") in new stack
> -- Called g2/7500
> -- Zap/32-1 is ringing
> -- Zap/32-1 answered SIP/8110-71ee
> -- Channel 0/1, span 2 got hangup
> -- Hungup 'Zap/32-1'
>   == Spawn extension (default, 7500, 1) exited non-zero on 'SIP/8110-71ee'
> Somebody would have some idea to help in this case me?
> Greatings
> Josué

Could you please post your zaptel.conf and zapata.conf ?
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Re: [Asterisk-Users] Problem: ringtones stop unexpectedly

2006-04-01 Thread Carlos A. Alfaro

I should've mentioned that before.  I've tried doing that and it has no
effect.  I've tried both upper and lower-case 'r's.

I've also tried a workaround that I thought would work, but it doesn't:
Answering the call and then using the playtones(ringing) command before
connecting to my cellphone.  

-Original Message-

Date: Sat, 1 Apr 2006 19:59:46 +0100
From: "Julian J. M." <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Problem: ringtones stop unexpectedly
whenmultiple channels are dialed
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1

Try adding 'r' to the dial options. According to "show application dial":

r- Indicate ringing to the calling party. Pass no audio to the
calling
   party until the called channel has answered.


exten => 3058472194,1,Dial(SIP/1035&SIP/[EMAIL PROTECTED],50, r)

Julian.

On 4/1/06, Carlos A. Alfaro <[EMAIL PROTECTED]> wrote:
>
>
>
> Hello Everyone.  I usually find my own solutions for problems but this
time,
> after several months, I've given up.
>
>
>
> My asterisk is set up so that incoming calls from my voip provider ring on
> both my sip extension and my cellphone at the same time.  When the system
> receives an incoming call, ringtones indicating that the call is being
> connected play normally for the first 5 seconds to the caller, but they
> suddenly stop as the call to my cellphone starts to make progress.  This
> causes some people to hang up, despite the fact that the call is still
being
> connected.  Callers who stay on the line are able to talk to me on either
> the sip extension or the cellphone once I pick up either one.
>
>
>
> I have tried a lot of workarounds like including a priority to answer the
> incoming call, invoke the playtones command before the dial command, but
> this doesn't seem to work either.  Can anyone replicate the problem?  Have
I
> ran into a bug?  I have pasted as much info as I deemed relevant; please
let
> me know if I'm missing something.  Thanks.


--

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End of Asterisk-Users Digest, Vol 21, Issue 2
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Re: [Asterisk-Users] H323 on way voice

2006-04-01 Thread isamar



Good luck. Try to switch between channel drivers.
Chan_oh323, chan_h323 and ooh323.
and remember to install the *exact* lib versions recommended on the 
readmes


May the force be with you...

Isamar


On Sat, 1 Apr 2006, Il Neofita wrote:


Hi,
I installed H323, however when I make a call from SIP Phone -> Asterisk H323
-> Provider H323 the provider can hear me, but I cannot hear nothing.
The asterisk is 1.2.6 with G729 license, and the asterisk is connect direct
to internet with a public IP.
Any thoughts?


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[Asterisk-Users] H323 on way voice

2006-04-01 Thread Il Neofita

Hi,I installed H323, however when I make a call from SIP Phone -> Asterisk H323 -> Provider H323 the provider can hear me, but I cannot hear nothing.The asterisk is 1.2.6 with G729 license, and the asterisk is connect direct to internet with a public IP.
Any thoughts?
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[Asterisk-Users] FreePBX on Debian

2006-04-01 Thread Christian Gröger


Hi
I got Asterix running on Debian Etch a few days ago. Version 1.2.6. 
Today i tried to install Freepbx and I got really confused. Do I really 
need that Zaptel stuff? It always prompted errors so i am now using 
mISDN -without errors, is there a module for freePBX for mISDN?
Then I tried installing that spandsp stuff, but makefile patching didn't 
work and i don't know what to do there manually... Next error was in 
that cdr-mysql-thing in asterisk-addons with much warnings during 
compiling :-/


Anyway, is there a good manual for installing FreePBX on debian? 
Something with typical debian-errors and stuff? That standard manual is 
so focussed on Suse :(

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Re: [Asterisk-Users] Re: no audio

2006-04-01 Thread Tom Vile

yes, it was fixed immediately following this bug.  Update your version
of Asterisk and you will be fine.

Also in AAH you should do the following in sip.conf

externip=your ISP external IP address
localnet=your internal network (ie) 192.168.1.0/255.255.255.0

remove nat=yes in sip.conf and add it in the extension setup.  There
is a line in there to change nat to yes.

Reload sip and you should be good to go.

On 4/1/06, hugolivude <[EMAIL PROTECTED]> wrote:
> Wierd timing - I'm struggling with exactly the same issue.  My problem
> was with ZAP - ZAP.  The phones ring, but no audio.  Turns out there's
> a bug with the version I'm running.  It has to do w/ the system date.
> When I changed my system date to 1-Jan-06, everything worked!!  Here's
> what I found from another posting:
>
> >this is a new bug which is submitted: http://bugs.digium.com/view.php?id=6349
> >change your system date to an older value. everything will work again.
>
> I'm hoping the bug is fixed in a more recent release build, but I
> haven't tried yet.
>
> Yours,
> Hugh
>
> On 4/1/06, Luis herrera <[EMAIL PROTECTED]> wrote:
> > Hi. I have a [EMAIL PROTECTED] setup at my home. My problems is with
> > phones outside my network. I call the extensions
> > without a problem, it rings but when they answer I
> > can't hear them and they can hear me.
> > I set up in the SIP.CONF
> > nat=yes
> >
> > I'm I missing any other setting or do I need a special
> > switch that support asterisk.
> > Thank you for your help.
> >
> >
> > __
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam protection around
> > http://mail.yahoo.com
> > ___
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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Re: [Asterisk-Users] G729 codec problems

2006-04-01 Thread Rudolf Ladyzhenskii

I am not. I have one license and use i channel.
It seems to detect the fact there are no more channels left and keeps
warning me about it in case I want to use more.

It is fine, but the warning is constant. All you see on Asterisk
console is running warning message.

Rudolf

On 4/2/06, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
> RumaTech wrote:
>
> > And it keeps running like that. Call usually come through OK. If i try
> > to use "show g729" command, it shows that all codecs are in use. Well,
> > this is fine, I am using one, but I do not want to see those warnings.
> > Once is quite enough. Those continuos warnings make it impossible to se
> > any other asterisk output. How does one turns them off?
>
> You can't make them stop except by not trying to use more channels than
> you have licenses for.
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Re: [Asterisk-Users] G729 codec problems

RumaTech wrote:

> And it keeps running like that. Call usually come through OK. If i try
> to use "show g729" command, it shows that all codecs are in use. Well,
> this is fine, I am using one, but I do not want to see those warnings.
> Once is quite enough. Those continuos warnings make it impossible to se
> any other asterisk output. How does one turns them off?

You can't make them stop except by not trying to use more channels than
you have licenses for.
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Re: [Asterisk-Users] G729 codec problems

2006-04-01 Thread RumaTech




Hi, all

Sorry, I was "out of action" for some time.

I am using Voipstnt to plae calls to USA/Canada and I bouhj 1 copy og G.729.
This was mainly to get one of the local Australians VoIP providers working.

Anyway, when I am trying to place calls to USA, it tries to use G.729 and I
am getting continuous warning:

20:56:20 WARNING[6620]: codec_g729.c:170 g729tolin_framein: Out of G.729 
Decoder Licenses!
20:56:20 WARNING[6620]: codec_g729.c:170 g729tolin_framein: Out of G.729 
Decoder Licenses!
20:56:20 WARNING[6620]: codec_g729.c:170 g729tolin_framein: Out of G.729 
Decoder Licenses!
20:56:20 WARNING[6620]: codec_g729.c:170 g729tolin_framein: Out of G.729 
Decoder Licenses!


And it keeps running like that. Call usually come through OK. If i try to 
use "show g729" command, it shows that all codecs are in use. Well, this is 
fine, I am using one, but I do not want to see those warnings. Once is quite 
enough. Those continuos warnings make it impossible to se any other asterisk 
output. How does one turns them off?


Thanks,
Rudolf






- Original Message - 
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial 
Discussion" 

Sent: Sunday, March 26, 2006 8:14 PM
Subject: Re: [Asterisk-Users] G729 codec problems



What sort of call path are you trying to get working?

Paul Hales
Technical Manager
AsteriskIT

- Original Message - 
From: "Rudolf Ladyzhenskii" <[EMAIL PROTECTED]>

To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Sunday, March 26, 2006 10:18 AM
Subject: [Asterisk-Users] G729 codec problems



Hi, all

I have a license for G.729A codec from Digium.

When asterisk starts it shows:
Jun 17 21:13:59 NOTICE[4040]: codec_g729.c:460 load_module: G.729
transcoding module Copyright (C) 1999-2005 Digium, Inc.
Jun 17 21:13:59 NOTICE[4040]: codec_g729.c:461 load_module: This
module is supplied under a commercial license granted by Digium, Inc.
Jun 17 21:13:59 NOTICE[4040]: codec_g729.c:462 load_module: Please see
the full license text supplied by the accompanying
Jun 17 21:13:59 NOTICE[4040]: codec_g729.c:463 load_module: "register"
utility, or ask for a copy from Digium.
  == G.729 Host-ID:

cc:20:a3:86:01:93:53:92:2c:37:ae:e7:ad:16:6e:f0:39:f6:88:4e

  == Found license 'G729-190B962C' providing 1 channels
  == Found total of 1 G.729 licenses
  == Registered translator 'g729tolin' from format g729 to slin, cost 20
  == Registered translator 'lintog729' from format slin to g729, cost 
115



All is fine, however when trying to make a call I am getting:
WARNING[4063]: codec_g729.c:170 g729tolin_framein: Out of G.729
Decoder Licenses!

No other calls are active.

Any ideas what is going on?

Thanks,
Rudolf
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[Asterisk-Users] vmail access problem

2006-04-01 Thread Ever Zalazar




Hi everybody..I have the follow problem with my 
vmail access:
 
http://voicemail.cheaphone.com/cgi-bin/vmail.cgi?action="">   
For example this is the address to access the voice mail of one customer. If 
that customer change the number for :
 
http://voicemail.cheaphone.com/cgi-bin/vmail.cgi?action="">   
He will access that user account and see the messages.HOw can I protect 
this?
 
 
Thanks
 
 
Ever
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[Asterisk-Users] channel.c:787 channel_find_locked: Avoided initial deadlock for '0x8446b50', 10 retries!

2006-04-01 Thread Il Neofita

I never so this error.I am using H323 with Asterisk 1.2.6 Any idea what can be the problem?
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Re: [Asterisk-Users] Install problem with res_snmp.so from current trunk (bug?)

Rich Adamson wrote:

> Is this worthy of opening a bug assuming the above comment is still
> valid?  Would the individual(s) maintaining res_snmp want to log into
> either of these internet accessible boxes to identify the root cause?

The module loader in trunk is undergoing changes that will eliminate
this problem very soon.
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RE: [Asterisk-Users] OT: Polycom IP501 and Speed Dials

2006-04-01 Thread Jeff Herring

http://www.voip-info.org/wiki/view/Polycom+reboot+hardphone+script

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Avi Miller
Sent: Saturday, April 01, 2006 4:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] OT: Polycom IP501 and Speed Dials


Mojo with Horan & Company, LLC wrote:
> if you reboot your phones from the asterisk server ie via cron or so, 
> that reboot script could potentially delete the phone-specific directory 
> xml before the sip message is sent

Sadly, that doesn't work -- the Polycoms store their directories locally 
as well and re-upload them on reboot.

Though, if you have a sample of that remote reboot script for the 
phones, I'd appreciate a copy. :)

cYa,
Avi

-- 
National Manager - Special Projects

< Melbourne / Sydney / Canberra / Hobart / London />
   2/340 Gore StreetT: +61 (0) 3 9486 0411
   Fitzroy, VIC F: +61 (0) 3 9486 0611
   3065 W: http://www.squiz.net/

.>> Open Source  - Own it  -  Squiz.net ./>
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[Asterisk-Users] Install problem with res_snmp.so from current trunk (bug?)

Just updated two fc3 systems running svn trunk. One updated, installed 
properly, and is working fine. The second box failed during the 'make 
install' process with:


/usr/lib/libnetsnmp.a(parse.o)(.text+0x275a): In function `unload_module':
: multiple definition of `unload_module'

res_snmp.o(.text+0x310):/usr/src/asterisk/res/res_snmp.c:102: first 
defined here
/usr/bin/ld: Warning: size of symbol `unload_module' changed from 66 in 
res_snmp.o to 284 in /usr/lib/libnetsnmp.a(parse.o)


collect2: ld returned 1 exit status
make[1]: *** [res_snmp.so] Error 1
make[1]: Leaving directory `/usr/src/asterisk/res'
make: *** [subdirs] Error 1


I noticed in the res/Makefile includes the following:

# NETsnmp has some difficulties on some platforms (conflict with 
unload_module)

# Until we figure out if the collission is version-specific or what to do
# we have disabled res_snmp on OS/X and *BSD

The two fc3 boxes have different versions of the /usr/lib/libnetsnmp
modules (since they are different sizes on the two boxes).

My work around was to simply comment out the res/Makefile steps to 
compile the res_snmp module, and to use 'noload res_snmp.so'.


Is this worthy of opening a bug assuming the above comment is still 
valid?  Would the individual(s) maintaining res_snmp want to log into 
either of these internet accessible boxes to identify the root cause?


Rich



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[Asterisk-Users] Re: no audio

2006-04-01 Thread hugolivude

Wierd timing - I'm struggling with exactly the same issue.  My problem
was with ZAP - ZAP.  The phones ring, but no audio.  Turns out there's
a bug with the version I'm running.  It has to do w/ the system date. 
When I changed my system date to 1-Jan-06, everything worked!!  Here's
what I found from another posting:

>this is a new bug which is submitted: http://bugs.digium.com/view.php?id=6349
>change your system date to an older value. everything will work again.

I'm hoping the bug is fixed in a more recent release build, but I
haven't tried yet.

Yours,
Hugh

On 4/1/06, Luis herrera <[EMAIL PROTECTED]> wrote:
> Hi. I have a [EMAIL PROTECTED] setup at my home. My problems is with
> phones outside my network. I call the extensions
> without a problem, it rings but when they answer I
> can't hear them and they can hear me.
> I set up in the SIP.CONF
> nat=yes
>
> I'm I missing any other setting or do I need a special
> switch that support asterisk.
> Thank you for your help.
>
>
> __
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> Tired of spam?  Yahoo! Mail has the best spam protection around
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[Asterisk-Users] chan-capi: Sending digits on a bri (isdn) d-channel

2006-04-01 Thread Raoul Bönisch

Dear asterisk users!

I want to control a hardware pbx with asterisk. What I need to do
this is being able to press "hold" which can be done with
capicommand(hold) and then send digits on a bri card which
connects to my asterisk computer. So far I use
Dial(CAPI/ISDN1/27:<>/bo,15) to do this. Are there better
ways? Note that these are not dtmf, I'm afraid.

I use an AVM Fritz!classic ISA card with the fcclassic kernel
module on linux 2.6.16, asterisk version 1.2.4 on Debian unstable.
To use the AVM card with asterisk I compiled the latest cvs tree of
chan-capi-cm. The hardware pbx is a T-Eumex 312 (product of the
german Telekom).

What I basically do is pretending asterisk is an original isdn
phone connected to the hardware pbx and it should send the proper
key presses that the hardware pbx understands. E.g. I'd like to
forward a call. Therefore asterisk would have to press "hold" when
a call is active, then dial the number of the phone the call should
be forwarded to, then press "*", "6", "1".

I use this extension to do it:

exten => 29,1,Answer
exten => 29,2,Wait(3)
exten => 29,3,Playback(echo-test)
exten => 29,4,Wait(1)
exten => 29,5,capicommand(hold)
exten => 29,6,Dial(CAPI/ISDN1/27:12/bo,15)
exten => 29,7,capicommand(retrieve)
exten => 29,8,Playback(echo-test)
exten => 29,9,Playback(Welcome)
exten => 29,10,Wait(5)
exten => 29,11,Hangup

Note, that in Dial(CAPI/ISDN1/27:12/bo,15), the 12 is the internal
number of the phone asterisk should forward the call to. Welcome and
echo-test are just for testing purposes. I can hear the first output
of echo-test when I call extension 29 and the call is properly put
on "hold". Then phone number 12 rings as expected. However when I
pick up phone number 12, the connections are hung up.

I derive this from the following output of asterisk -r -vvv:

-- CONNECT_IND
(PLCI=0x101,DID=29,CID=14,CIP=0x4,CONTROLLER=0x1)
  == Started pbx on channel CAPI/ISDN1/29-20
-- Executing Answer("CAPI/ISDN1/29-20", "") in new stack
-- Executing Wait("CAPI/ISDN1/29-20", "3") in new stack
Apr  2 00:16:41 WARNING[11926]: channel.c:1591
ast_waitfor_nandfds: Thread -1230230608 Blocking
'CAPI/ISDN1/29-20', already blocked by thread 0 in procedure
(null)
-- Executing Playback("CAPI/ISDN1/29-20", "echo-test") in new
stack
-- Playing 'echo-test' (language 'de')
-- Executing Wait("CAPI/ISDN1/29-20", "1") in new stack
-- Executing capiCommand("CAPI/ISDN1/29-20", "hold") in new
stack
-- capiCommand: 'hold' '(null)'
   > ISDN1: sent HOLD for PLCI=0x101
-- Executing Dial("CAPI/ISDN1/29-20",
"CAPI/ISDN1/27:12/bo|15") in new stack
   > data = ISDN1/27:12/bo
   > capi request for interface 'ISDN1'
  == ISDN1: Call CAPI/ISDN1/12-21 with B3 overlap (pres=0x00,
ton=0x41)
-- Called ISDN1/27:12/bo
Apr  2 00:16:48 WARNING[11926]: channel.c:1591
ast_waitfor_nandfds: Thread -1230230608 Blocking
'CAPI/ISDN1/12-21', already blocked by thread 0 in procedure
(null)
-- ISDN1: received CONNECT_CONF PLCI = 0x201
-- ISDN1: PLCI=0x101 put onhold
-- CAPI/ISDN1/12-21 is making progress passing it to
CAPI/ISDN1/29-20
-- CAPI/ISDN1/12-21 is ringing
-- ISDN1: attempting ALERT in state 10
-- CAPI/ISDN1/12-21 answered CAPI/ISDN1/29-20
   > ISDN1: using PLCI=0x101 for retrieve
   > ISDN1: sent RETRIEVE for PLCI=0x101
-- Attempting native bridge of CAPI/ISDN1/29-20 and
CAPI/ISDN1/12-21
-- ISDN1: activehangingup (cause=0)
  == Spawn extension (from-tk, 29, 6) exited non-zero on
'CAPI/ISDN1/29-20'
  == ISDN1: Interface cleanup PLCI=0x101
  == ISDN1: Interface cleanup PLCI=0x201


Note the statement "ISDN1: activehangingup (cause=0)" which is
what reports the call being hung up. This is not what I expect.
The call should have been forwarded to phone number 12 and not
hung up. How can I avoid this?

Thanks in advance.

Raoul

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[Asterisk-Users] no audio

2006-04-01 Thread Luis herrera

Hi. I have a [EMAIL PROTECTED] setup at my home. My problems is with
phones outside my network. I call the extensions
without a problem, it rings but when they answer I
can't hear them and they can hear me.
I set up in the SIP.CONF
nat=yes

I'm I missing any other setting or do I need a special
switch that support asterisk.
Thank you for your help.


__
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Re: [Asterisk-Users] TO have ringing tone instead MOH

2006-04-01 Thread Melcon Moraes

Can I ask you why?

[]'s
MM

 -Original Message-
From:   Alberto Sagredo <[EMAIL PROTECTED]>
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Cc: 
Sent:  Sat, 01 Apr 2006 20:54:33 +0200
Delivered:  Sat,  01 Apr 2006 15:56:25 
Subject:[Asterisk-Users] TO have ringing tone instead MOH

I need to avoid MOH on my asterisk box, so i need to have a ringing tone 
when attendant transfer is made, or a call is on hold..

Is there any way to do that.

I did not see a simple way to do that.

Regards

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E-mail classificado pelo Identificador de Spam Inteligente Terra.
Para alterar a categoria classificada, visite
http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz&_l=1,1143917785.931094.20477.ambrose.hst.terra.com.br,3471,Des15,Des15

 --Original Message Ends--

-- 
Melcon Moraes <[EMAIL PROTECTED]>

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RE: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?

Check the musiconhold.conf.sample in the asterisk/configs directory.
That will tell you what you need to know.

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: 01 April 2006 16:41
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?

How did you switch from native to mpg123 on 1.2.x?  That's what I can't
figure out.


On 4/1/06, Lee Archer <[EMAIL PROTECTED]> wrote:
> Has anyone else had a problem with asterisk creating multiple threads?
> I'm still testing but I've move from native to mpg123 for the machine 
> with the problem and the problem hasn't come back.
>
> Lee
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> Sent: 01 April 2006 15:07
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?
>
> Ok this is great... but I just noticed this morning while doing some 
> tests that asterisk seems to start a new stream for every caller
> With mpg123 it would just start one and all calls would hear the same
> stream.Unless something was seriously lagging, my test calls this
> morning all were in different spots in the hold music.   Isn't this
> less efficient?
>
>
> On 4/1/06, Melcon Moraes <[EMAIL PROTECTED]> wrote:
> > You don't have to use it in newer versions. Get your mp3, ant 
> > convert to slin format with sox.
> >
> > Ex: sox -V file.mp3 [-c1] file.slin
> >
> > -V: just to show you what's going on
> > -c1: convert to 1 channel, if your mp3 is stereo
> >
> > Then edit your musiconhold.conf like this:
> >
> > [native]
> > mode=files
> > directory=/var/lib/asterisk/moh-native
> >
> > and you'll have a nice native streaming. You can convert your stuff 
> > to
>
> > another formats, like "sox file.mp3 [-c1] file.gsm" or "sox file.mp3

> > [-c1] file.ul" and let asterisk decide which one best fits given
> channel.
> >
> > []'s
> > MM
> >
> >  -Original Message-
> > From:   "Lee Archer" <[EMAIL PROTECTED]>
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> > Cc:
> > Sent:  Sat, 1 Apr 2006 10:34:42 +0100
> > Delivered:  Sat,  01 Apr 2006 06:28:16 Subject:[Asterisk-Users] 
> > 1.2.6 doesn't use mpg123?
> >
> > I use mpg123 for streaming but I can't get it to compile under 
> > SuSe10 and x86_64 CPU.  Does anyone have any recommendations for 
> > other programs that allow streaming and will compile on this arch?
> >
> > Regards
> >
> > Lee
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> > Sent: 31 March 2006 22:36
> > To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
> > Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] 1.2.6 doesn't use mpg123?
> >
> > > >
> > > And isn't mpg123 ( or some replacement ) required when using a 
> > > stream for MOH I couldn't get streaming to work without it in 1.2?
> >
> > Yes.. mpg123 is required for streaming... I had it working in
1.0.9...
> > though have not tried in 1.2.
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> > Asterisk-Users mailing list
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> >
> >
> > ###
> >
> > This message has been scanned by F-Secure Anti-Virus for Microsoft
> Exchange.
> > For more information, connect to http://www.f-secure.com/ 
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> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
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> >
> > E-mail classificado pelo Identificador de Spam Inteligente Terra.
> > Para alterar a categoria classificada, visite 
> > http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz&;
> > _l
> > =1,1143884189.96596.438.aldavila.hst.terra.com.br,5146,Des15,Des15
> >
> >
> >  --Original Message Ends--
> >
> > --
> > Melcon Moraes <[EMAIL PROTECTED]>
> >
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Re: [Asterisk-Users] *.conf generator

Other thing I'm thinking... why are you running astlinux?  Asterisk
really isn't that hard to install..

make; make install; make config


On 4/1/06, Matt <[EMAIL PROTECTED]> wrote:
> I don't understand your question.   You don't want to generate the
> config files by hand, but yet you can't use FreePBX?   Why can FreePBX
> not generate the conf files and then you go on to use astlinux?
> FreePBX should run on any linux distribution.
>
>
> On 4/1/06, mustardman29 <[EMAIL PROTECTED]> wrote:
> >
> > Is there a good free *.conf generator out there.  Manual configuration is
> > just too tedious.  I run Astlinux so a lot of the GUI's such as AMP
> > (FreePBX)are not an option either.
> >
> > I used to use IPManager which did a great job but that project has been
> > discontinued :(.
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> >
>
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Re: [Asterisk-Users] *.conf generator

I don't understand your question.   You don't want to generate the
config files by hand, but yet you can't use FreePBX?   Why can FreePBX
not generate the conf files and then you go on to use astlinux? 
FreePBX should run on any linux distribution.


On 4/1/06, mustardman29 <[EMAIL PROTECTED]> wrote:
>
> Is there a good free *.conf generator out there.  Manual configuration is
> just too tedious.  I run Astlinux so a lot of the GUI's such as AMP
> (FreePBX)are not an option either.
>
> I used to use IPManager which did a great job but that project has been
> discontinued :(.
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Re: [Asterisk-Users] TO have ringing tone instead MOH

Use r instead of m in your transfer contexts.. and umm if you really
want 'ringing' when someone is on hold... just put a sound file with
'ringing' in it in the moh directory.  Otherwise, you can disable MOH
and you will just have silence when someone is on hold.


On 4/1/06, Alberto Sagredo <[EMAIL PROTECTED]> wrote:
> I need to avoid MOH on my asterisk box, so i need to have a ringing tone
> when attendant transfer is made, or a call is on hold..
>
> Is there any way to do that.
>
> I did not see a simple way to do that.
>
> Regards
>
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RE: [Asterisk-Users] AGI hangup problem

2006-04-01 Thread Branko Samardzic

Well that's fine but then I don't have any channel variable info I might
have from channel otherwise. I am catchning exception now but I am wondering
if there is some more appropriate way for Asterisk to handle hangup (to
stick around for some more time just in case someone wants to pull any
channel info after hangup).


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Stefan
Reuter
Sent: Saturday, April 01, 2006 2:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AGI hangup problem


Branko Samardzic wrote:
> I have problem with Asterisk.
> [sendCommand]=EXEC "DIAL" "IAX2/somehost/somenumber|10"
> [readReply]=200 result=-1
> [sendCommand]=GET VARIABLE "ANSWEREDTIME"
> 1086422 [Thread-7] ERROR - establishConnection: exec encountered exception
> net.sf.asterisk.fastagi.AGIHangupException: Channel was hung up.
> This means that AGI is not capable of extracting data about call
performed.
> Is there any workaround?

just catch the exception and go on...


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Re: [Asterisk-Users] Iaxmodem speed limit?

2006-04-01 Thread Lee Howard


Carlos Chavez wrote:


I just installed Hylafax with Iaxmodem and I am not getting good
results when receiving faxes.  I can see that the modem is reporting the
following:

Mar 31 16:19:08 pbxoficina FaxGetty[5377]: MODEM Supports 2400 bit/s
Mar 31 16:19:08 pbxoficina FaxGetty[5377]: MODEM Supports 4800 bit/s
Mar 31 16:19:08 pbxoficina FaxGetty[5377]: MODEM Supports 7200 bit/s
Mar 31 16:19:08 pbxoficina FaxGetty[5377]: MODEM Supports 9600 bit/s
Mar 31 16:19:08 pbxoficina FaxGetty[5377]: MODEM Supports 12000 bit/s
Mar 31 16:19:08 pbxoficina FaxGetty[5377]: MODEM Supports 14400 bit/s

Is there a way to limit the speed of Hylafax to 7200 bits/s to get more
reliable results?



I have enabled V.17 in spandsp which causes these FaxGetty "MODEM 
Supports" messages you quote here.  You can disable any particular 
speeds with the Class1RMQueryCmd and Class1TMQueryCmd options:


 Class1RMQueryCmd:  "!24,48,72,96"
 Class1TMQueryCmd:  "!24,48,72,96"

Will disable V.17 in sending and receiving.  My iaxmodem config file for 
HylaFAX suggests using this Class1RMQueryCmd option but not the 
Class1TMQueryCmd (so V.17 works in sending but not receiving).


That said, eliminating any speed may not get you any more "reliable" 
results.  If you're having problems post the HylaFAX session logs here, 
and we'll try to go from there.


Lee.

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Re: [Asterisk-Users] Incorrect CDR results

2006-04-01 Thread Andrew Kohlsmith

On Saturday 01 April 2006 14:09, Michael Welter wrote:
> When I look at my CDR data for calls to NuFone, the billsec for each
> call is 14 seconds or less.  When I look at my NuFone account, the
> billsec has normal call lengths.

Are you transfering off of your asterisk box?  IAX2 by default will try to 
minimize the connection path by dropping out servers which don't need to be 
there.  For you nufone iax2 peer, make sure you specify 'notransfer=yes' and 
reload iax.conf.

-A.
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[Asterisk-Users] *.conf generator

2006-04-01 Thread mustardman29

 
Is there a good free *.conf generator out there.  Manual configuration is
just too tedious.  I run Astlinux so a lot of the GUI's such as AMP
(FreePBX)are not an option either.

I used to use IPManager which did a great job but that project has been
discontinued :(.
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[Asterisk-Users] Incorrect CDR results

2006-04-01 Thread Michael Welter

When I look at my CDR data for calls to NuFone, the billsec for each 
call is 14 seconds or less.  When I look at my NuFone account, the 
billsec has normal call lengths.


So it seems that the billing on the Asterisk system terminates after 
about 14 seconds.  The calls come in on an IAX connection and go out to 
NuFone on IAX.  Are these calls bridging away from the Asterisk server? 
 How can I get accurate billing data?


I tried to Google the archives but I'm still getting "page not found".

Thanks

--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
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Re: [Asterisk-Users] AGI hangup problem

2006-04-01 Thread Stefan Reuter

Branko Samardzic wrote:
> I have problem with Asterisk.
> [sendCommand]=EXEC "DIAL" "IAX2/somehost/somenumber|10"
> [readReply]=200 result=-1
> [sendCommand]=GET VARIABLE "ANSWEREDTIME"
> 1086422 [Thread-7] ERROR - establishConnection: exec encountered exception
> net.sf.asterisk.fastagi.AGIHangupException: Channel was hung up.
> This means that AGI is not capable of extracting data about call performed.
> Is there any workaround?

just catch the exception and go on...



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Re: [Asterisk-Users] Problem: ringtones stop unexpectedly when multiple channels are dialed

2006-04-01 Thread Julian J. M.

Try adding 'r' to the dial options. According to "show application dial":

r- Indicate ringing to the calling party. Pass no audio to the calling
   party until the called channel has answered.


exten => 3058472194,1,Dial(SIP/1035&SIP/[EMAIL PROTECTED],50, r)

Julian.

On 4/1/06, Carlos A. Alfaro <[EMAIL PROTECTED]> wrote:
>
>
>
> Hello Everyone.  I usually find my own solutions for problems but this time,
> after several months, I've given up.
>
>
>
> My asterisk is set up so that incoming calls from my voip provider ring on
> both my sip extension and my cellphone at the same time.  When the system
> receives an incoming call, ringtones indicating that the call is being
> connected play normally for the first 5 seconds to the caller, but they
> suddenly stop as the call to my cellphone starts to make progress.  This
> causes some people to hang up, despite the fact that the call is still being
> connected.  Callers who stay on the line are able to talk to me on either
> the sip extension or the cellphone once I pick up either one.
>
>
>
> I have tried a lot of workarounds like including a priority to answer the
> incoming call, invoke the playtones command before the dial command, but
> this doesn't seem to work either.  Can anyone replicate the problem?  Have I
> ran into a bug?  I have pasted as much info as I deemed relevant; please let
> me know if I'm missing something.  Thanks.
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[Asterisk-Users] TO have ringing tone instead MOH

2006-04-01 Thread Alberto Sagredo

I need to avoid MOH on my asterisk box, so i need to have a ringing tone 
when attendant transfer is made, or a call is on hold..


Is there any way to do that.

I did not see a simple way to do that.

Regards

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[Asterisk-Users] Problem: ringtones stop unexpectedly when multiple channels are dialed

2006-04-01 Thread Carlos A. Alfaro









Hello Everyone.  I usually find my own solutions for
problems but this time, after several months, I’ve given up.

 

My asterisk is set up so that incoming calls from my voip
provider ring on both my sip extension and my cellphone at the same time. 
When the system receives an incoming call, ringtones indicating that the call
is being connected play normally for the first 5 seconds to the caller, but
they suddenly stop as the call to my cellphone starts to make progress. 
This causes some people to hang up, despite the fact that the call is still
being connected.  Callers who stay on the line are able to talk to me on
either the sip extension or the cellphone once I pick up either one.

 

I have tried a lot of workarounds like including a priority
to answer the incoming call, invoke the playtones command before the dial
command, but this doesn’t seem to work either.  Can anyone replicate
the problem?  Have I ran into a bug?  I have pasted as much info as I
deemed relevant; please let me know if I’m missing something. 
Thanks.

 

Carlos

 

 

This is how I set up my extensions.conf to dial two channels
(my sip extension and my cellphone) when asterisk receives an incomming
call.  

 

EXTENSIONS.CONF:

 

[incoming]

exten =>
3058472194,1,Dial(SIP/1035&SIP/[EMAIL PROTECTED],50)

exten => 3058472194,2,Wait(2)

exten => 3058472194,3,voicemail(u1000)

exten => 3058472194,103,voicemail(b1000)

 

 

 

 

CONSOLE OUTPUT FOR THE INCOMING CALL:

 

asterisk*CLI>

    -- Executing
Dial("SIP/3058472194-ff33",
"SIP/1035&SIP/[EMAIL PROTECTED]|50") in new stack

    -- Called 1035

    -- Called [EMAIL PROTECTED]

    -- SIP/1035-21d1 is ringing

    -- SIP/richmedium-625f is ringing

    -- SIP/richmedium-625f is making progress
passing it to
SIP/3058472194-ff33
< (Ringtones stop at this point)

    -- SIP/richmedium-625f answered
SIP/3058472194-ff33   
 

    -- Attempting native bridge of
SIP/3058472194-ff33 and SIP/richmedium-625f

  == Spawn extension (internal, 3058472194, 1) exited
non-zero on 'SIP/3058472194-ff33'

    -- Executing
Hangup("SIP/3058472194-ff33", "") in new stack

  == Spawn extension (internal, h, 1) exited non-zero
on 'SIP/3058472194-ff33'

 

 

SIP.CONF:

 

register =>
[EMAIL PROTECTED]:shh:[EMAIL PROTECTED]

 

[3058472194bv]

type=peer

user=phone

context=incoming

host=sip.broadvoice.com

fromdomain=sip.broadvoice.com

fromuser=3058472194

secret=s

username=3058472194

insecure=very

authname=3058472194

nat=no

dtmfmode=rfc2833

 

[richmedium]

type=friend

username=car3423

secret=s

host=64.135.90.5

dtmfmode=rfc2833

disallow=all

;allow=g729

;allow=g726

allow=ulaw

;allow=ilbc

;allow=gsm

context=disconnected

insecure=very

 

 

 

MY SYSTEM:

 

[EMAIL PROTECTED] ~]# uname -a

Linux 2.6.9-22.0.2.EL #1 Tue Jan 17 06:51:40 CST 2006 i686
i686 i386 GNU/Linux

 

 

 

 

ASTERISK VERSION:

 

Asterisk 1.2.4

 

 

 

CONSOLE DEBUG OUTPUT:

 

Too big for this posting, had to remove
it, but can paste on a followup.

 

 






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[Asterisk-Users] AGI hangup problem

2006-04-01 Thread Branko Samardzic

I have problem with Asterisk.
[sendCommand]=EXEC "DIAL" "IAX2/somehost/somenumber|10"
[readReply]=200 result=-1
[sendCommand]=GET VARIABLE "ANSWEREDTIME"
1086422 [Thread-7] ERROR - establishConnection: exec encountered exception
net.sf.asterisk.fastagi.AGIHangupException: Channel was hung up.
This means that AGI is not capable of extracting data about call performed.
Is there any workaround?

Cheers,
Branko

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[Asterisk-Users] Free Software/Open Source Telephony-Summit 2006

 Free Software/Open Source Telephony-Summit 2006

  Tuesday,  May 2nd 2006

Wiesbaden, Germany

For the third time the German Unix User Group (GUUG - www.guug.de)
organizes the Free Software/Open Source Telephony-Summit, an
international workshop and technical conference for developers and
users of Free Software/Open Source telephony applications and for
those who want to know more about what the Free Software/Open Source
world has to offer in the telephony and VoIP sector.

For the first time the event will this year be held in association
with LinuxTag (www.linuxtag.org), Europe's most important Open Source
event. This way participants of the Telephony-Summit can profit from
other LinuxTag events to deepen their knowledge.

The all-day conference with two parallel tracks will take place in
Wiesbaden, Germany. In the evening a social event is planned and
included in the price of 360 EUR. (Students only pay 120 EUR.) Members
of the German Unix User Group will get a 10% discount. If you register
before April 13th 2006, you'll get an additional 10% early bird discount.

Conference participants can visit the LinuxTag exhibition and talks
on the next days, which take place nearby. You can also book additional
tutorials.

The conference program includes presentations from many Open Source
projects like Asterisk, Ekiga, and OpenPBX. There are talks about
SIP security, RADIUS and LDAP integration as well as reports about
efforts to create carrier-class telephony switches.

On the weekend before the conference a workshop will be held. This
workshop is for invited Free Software/Open Source developers who can
meet their peers and work together on new versions of their software,
do interoperability tests and discuss their projects. Most developers
will also be at the conference, so this is your chance to talk to the
developers of your favourite projects.

For more information and registration see the website at:
http://www.linuxtag.org/2006/en/besucher/konferenzen/3rd-telephony-summit.html

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RE: Re: [Asterisk-Users] Routing SIP calls via URI

2006-04-01 Thread Shad Mortazavi

Dear Group,

I was able to fix this problem;

The solution was to use a prefix to dial out. 

The next challenge was to send the SIP Domain over IAX2!. I found that
if I included @SIPDOMAIN it would break the IAX2 communications.

exten => _6.,1,Dial(IAX2/bxx:[EMAIL PROTECTED]/[EMAIL PROTECTED]),
breakes because @SIPDOMAIN is treated as the target context. You also
can not include @Context after the @SIPDOMAIN.

I created a new variable DS which was a concatenation of EXTEN and
SIPDOMAIN separated by % and not @ and I was now able to pass this over
IAX2;

DS = EXTEN%SIPDOMAIN.

exten => _6.,1,Dial(IAX2/bxx:[EMAIL PROTECTED]/${DS}).

At the other end I used the CUT command and substring facilities in
Asterisk to split DS by the % eliminator; I re-formed a new variable
which was 

DS = [EMAIL PROTECTED]

I can now pass calls from my internal Asterisk server to my external
Asterisk server using IAX2 and then call any external VoIP number.

Warm Regards

Shad Mortazavi
--
Nexus Group Technical Manager
n|m Nexus Management Inc

-Original Message-
From: Shad Mortazavi 
Sent: Thursday, March 30, 2006 10:30 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Routing SIP calls via URI

Dear Group;

I can confirm that I have read through the three examples in
www.voip-info.org. 

These examples are excellent and address a couple of the questions. I
have IAX2 working between several asterisk servers on our VPN and
between the DMZ and our LAN. 

Also

exten => shad,1,Dial(IAX2/bxx:[EMAIL PROTECTED]/${EXTEN})

This answers part of the question;

However what I want to do is to send any outbound sip calls via our
external SIP server.

i.e;
 VPN  LANIAX2DMZ  Internet
Internal UA <---> Internal (*) <--> External (*)<-->
ExternalUA

We have an extensive internal dial plan, X dial the UK, Y dial USA, 1XXX
for Voicemail, 2xxx for Meetme, etc. 

Do I need to setup a prefix to dial the internet? And then route all
calls to the External(*) based on this prefix?

Thanks

Shad Mortazavi
--
Nexus Group Technical Manager
n|m Nexus Management Inc


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Re: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?

How did you switch from native to mpg123 on 1.2.x?  That's what I
can't figure out.


On 4/1/06, Lee Archer <[EMAIL PROTECTED]> wrote:
> Has anyone else had a problem with asterisk creating multiple threads?
> I'm still testing but I've move from native to mpg123 for the machine
> with the problem and the problem hasn't come back.
>
> Lee
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> Sent: 01 April 2006 15:07
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?
>
> Ok this is great... but I just noticed this morning while doing some
> tests that asterisk seems to start a new stream for every caller
> With mpg123 it would just start one and all calls would hear the same
> stream.Unless something was seriously lagging, my test calls this
> morning all were in different spots in the hold music.   Isn't this
> less efficient?
>
>
> On 4/1/06, Melcon Moraes <[EMAIL PROTECTED]> wrote:
> > You don't have to use it in newer versions. Get your mp3, ant convert
> > to slin format with sox.
> >
> > Ex: sox -V file.mp3 [-c1] file.slin
> >
> > -V: just to show you what's going on
> > -c1: convert to 1 channel, if your mp3 is stereo
> >
> > Then edit your musiconhold.conf like this:
> >
> > [native]
> > mode=files
> > directory=/var/lib/asterisk/moh-native
> >
> > and you'll have a nice native streaming. You can convert your stuff to
>
> > another formats, like "sox file.mp3 [-c1] file.gsm" or "sox file.mp3
> > [-c1] file.ul" and let asterisk decide which one best fits given
> channel.
> >
> > []'s
> > MM
> >
> >  -Original Message-
> > From:   "Lee Archer" <[EMAIL PROTECTED]>
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> > Cc:
> > Sent:  Sat, 1 Apr 2006 10:34:42 +0100
> > Delivered:  Sat,  01 Apr 2006 06:28:16 Subject:[Asterisk-Users] 1.2.6
> > doesn't use mpg123?
> >
> > I use mpg123 for streaming but I can't get it to compile under SuSe10
> > and x86_64 CPU.  Does anyone have any recommendations for other
> > programs that allow streaming and will compile on this arch?
> >
> > Regards
> >
> > Lee
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> > Sent: 31 March 2006 22:36
> > To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
> > Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] 1.2.6 doesn't use mpg123?
> >
> > > >
> > > And isn't mpg123 ( or some replacement ) required when using a
> > > stream for MOH I couldn't get streaming to work without it in 1.2?
> >
> > Yes.. mpg123 is required for streaming... I had it working in 1.0.9...
> > though have not tried in 1.2.
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> >
> >
> > ###
> >
> > This message has been scanned by F-Secure Anti-Virus for Microsoft
> Exchange.
> > For more information, connect to http://www.f-secure.com/
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> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > E-mail classificado pelo Identificador de Spam Inteligente Terra.
> > Para alterar a categoria classificada, visite
> > http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz&_l
> > =1,1143884189.96596.438.aldavila.hst.terra.com.br,5146,Des15,Des15
> >
> >
> >  --Original Message Ends--
> >
> > --
> > Melcon Moraes <[EMAIL PROTECTED]>
> >
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Re: [Asterisk-Users] Dial cmd has delay for the last dialed number on FXO


The * server one TDM04B card and my dialplan:

exten => 080.,1,Dial(Zap/g1/${EXTEN})

All four FXO ports has group=1 in zapata.conf

After dialing 0800012345 from a FXS extension, with one DTMF detector 
tapped on the line, I found * dialed 0,8,0,0,0,1,2,3,4 in even interval, 
delay longer, and then the last 5. Sometimes this behavior causes PSTN 
to repsond with "the number you dialed is non-existing".


Best guess given the above... asterisk is probably detecting a tone or 
noise coming from the central office, and/or, detecting tip-ring open or 
polarity reversal during the dialing. (I'm not familiar with your 
country telco standards, so this might not be all that helpful.)

Try...
 busydetect=yes
 busycount=6
 callprogress=no
towards the top of your zapata.conf statements.

If those don't have any impact, post a piece of your zapata.conf file.

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Re: [Asterisk-Users] Newbie question - sip.conf incoming contexts


I've been struggling with the documentation for months on this simple
subject...
I still have not been able to get this concept down...


I have 3 sip accounts (PSTN DID's) that come into my asterisk box
and give me phone service from my itsp via SIP.
I for the life of me have not been able to figure out how to get them to
come in to 3 seperate contexts!

This must be simple but I am missing the point.
All 3 accounts need a register line (I think) in order to work.

The register lines work great but I have not been able to figure out
how to get the other two lines to come into another seperate inbound
context that I have defined other than the one that is specified
in the [general] section of sip.conf

The /extension number does not do the trick for me

I wuld like for these incoming lines (from the same itsp) to truly
land in one of 3 seperate starting contexts in my dialplan based
on what phone number (account) they are.

Thank you very much for your help... this must be simple
but I have not really figured it out in several months of playing
around and reading

I've figured out a TON of other complex things, but this simple
incoming context thing has me a bit stumped.

I've tried a few things in my sip peer like
register=yes which was suggested on a web site but it does not work.

I also tried maing the peer name match the account (phone number)
of the sip account and that did not do it either.

The peers work fine as outgoing but I've not figured out how to make them
work for incoming as my sip itsp requires that I 'register' for inbound
calls.


What version of asterisk? (been lots of changes happening to the sip 
code over the last year)


Have you looked at the sample configs in /usr/src/asterisk/configs?

It would be far more helpful if you'd post your register statements and 
each of the sip contexts from sip.conf.  Might also include the section 
of your dialplan that each of the sip.conf contexts refer to.


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Re: [Asterisk-Users] Asterisk box with unreliable ping/latency



First of all, I should say that I’ve been running Asterisk on a Fedora 
Core 3 box since last May, but decided to do a reinstallation of 
everything due to some problems we’ve had with echos during 
conversations (100% SIP based, so no ZAP echos). We are talking about a 
low-volume installation running off an Intel Celeron 2.53 GHz box, 512 
MB RAM and some no-name MB 
(https://www.legendmemory.com/modules.php?op=modload&name=News&file=article&sid=412 
). 
The internet connection is DSL based, no NAT, eight IP addresses.


 

The problems continued. I did some tests on the DSL line, which itself 
seemed fine. But when I connected Asterisk to the network, suddenly ping 
became unstable. We are talking about ping varying between 20 ms and 600 
ms over a connection that usually should result in 20 ms ping. So I 
decided to eliminate the chance of increased latency due to traffic, 
bypassed the switch and hooked the Asterisk box up directly to the 
router. I even disabled Asterisk itself. Still, the problem persisted. 
It’s worth mentioning that the latency to the router itself was fine all 
the time, we’re just referring to the latency of the Asterisk box.


 

So I started installing [EMAIL PROTECTED] on a spare computer, about the same 
specs, only a different motherboard 
(http://www.dealtime.com/xPF-Asrock_MB_PM800_ASROCK_P4VM800_RTL_P4VM800). 
Same issues. At this very moment, I was left with a suspicion that there 
might be something with the DSL network, so I contacted the DSL provider 
and had them run tests on the line, which all came out just fine. I 
switched DSL modem/router, still same problems.


 

I then decided to do installations of different Linux distros (no 
Asterisk installations) on the spare computer, and did some interesting 
observings:


- Debian: Better, only occasional (a few percentage of the pings 
performed were above normal, and then only 200 ms above normal, compared 
to Centos/Asterisk where 30-40% were above normal). Still not 100%


- Mandriva: Perfect latency all the way

- Centos (base installation): Problems equal to the ones described earlier

 

I did some further testing, and on a P4 3,0 GHz, 512 MB RAM and a 
motherboard which I don’t know the maker /chipet, the CentOS 
installation came out with just a small percentage lag in latency.


 

I have checked for IRQ errors, and there seems to be no conflicts. I do 
see the network sharing IRQ with the USB bus, but this is common on 
motherboards with everything integrated.


 

I would assume this can be defined as off topic, as it clearly does not 
relate to Asterisk in particular. However, I am writing this in the hope 
of that someone might have had similar issues before and possibly been 
able to solve the problem. Google has not given me much on the subject. 
It’s not that I don’t mind putting together a new box with separate and 
higher level of quality components, but I’d much rather learn something 
from this experience than giving up ;)


You've provide a significant amount of general information, but nothing 
specific enough to guess at what might be going on.


Some things you might want to consider or try (at least to eliminate 
possibilities) include:
1. run Ethereal and trace the packets to/from your asterisk box, looking 
for anomalies in response times, layer-3 routing issues, etc.
2. run 'netstat -rn' to ensure the path from asterisk to the default 
gateway is exactly what you expect.
3. ensure the duplex setting of the nic card matches whatever that is 
connected to (eg, switch, hub). Don't depending on auto-negotiation 
functions working correctly.
4. run 'top' to ensure some other process isn't consuming processor 
cycles unexpectedly.


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RE: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?

Has anyone else had a problem with asterisk creating multiple threads?
I'm still testing but I've move from native to mpg123 for the machine
with the problem and the problem hasn't come back.

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: 01 April 2006 15:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?

Ok this is great... but I just noticed this morning while doing some
tests that asterisk seems to start a new stream for every caller 
With mpg123 it would just start one and all calls would hear the same
stream.Unless something was seriously lagging, my test calls this
morning all were in different spots in the hold music.   Isn't this
less efficient?


On 4/1/06, Melcon Moraes <[EMAIL PROTECTED]> wrote:
> You don't have to use it in newer versions. Get your mp3, ant convert 
> to slin format with sox.
>
> Ex: sox -V file.mp3 [-c1] file.slin
>
> -V: just to show you what's going on
> -c1: convert to 1 channel, if your mp3 is stereo
>
> Then edit your musiconhold.conf like this:
>
> [native]
> mode=files
> directory=/var/lib/asterisk/moh-native
>
> and you'll have a nice native streaming. You can convert your stuff to

> another formats, like "sox file.mp3 [-c1] file.gsm" or "sox file.mp3 
> [-c1] file.ul" and let asterisk decide which one best fits given
channel.
>
> []'s
> MM
>
>  -Original Message-
> From:   "Lee Archer" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"

> Cc:
> Sent:  Sat, 1 Apr 2006 10:34:42 +0100
> Delivered:  Sat,  01 Apr 2006 06:28:16 Subject:[Asterisk-Users] 1.2.6 
> doesn't use mpg123?
>
> I use mpg123 for streaming but I can't get it to compile under SuSe10 
> and x86_64 CPU.  Does anyone have any recommendations for other 
> programs that allow streaming and will compile on this arch?
>
> Regards
>
> Lee
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> Sent: 31 March 2006 22:36
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
> Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] 1.2.6 doesn't use mpg123?
>
> > >
> > And isn't mpg123 ( or some replacement ) required when using a 
> > stream for MOH I couldn't get streaming to work without it in 1.2?
>
> Yes.. mpg123 is required for streaming... I had it working in 1.0.9...
> though have not tried in 1.2.
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>
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>
> --
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>
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[Asterisk-Users] Refer Accountcode Bug

2006-04-01 Thread Elton Machado

I would like to know if there are any turn around for accountcode missing in
a refer request, I see the bug still marked as avaible and I would like to
know if in mean time if there are any solution avaible for it. 


Best Regards, 

Elton 


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Re: [Asterisk-Users] Confused on Agents and Queues

> >
> > However, anyone have a good way to log the agent out without having
> > them enter their agent ID and then have to hit # for the new
> > extension?
> >
>
>  There are a couple of ways listed here in the Wiki:
>
>  
> http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AgentCallbackLogin

Already viewed that.  And there aren't any good wasy to log an agent
out if the agent is using a unique agent id (not extension number) to
login.
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RE: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?

I want to stream shoutcast etc. but mpg123 won't compile.  I use native
moh with files but it won't work with streams.

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Melcon
Moraes
Sent: 01 April 2006 14:33
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?

You don't have to use it in newer versions. Get your mp3, ant convert to
slin format with sox. 

Ex: sox -V file.mp3 [-c1] file.slin

-V: just to show you what's going on
-c1: convert to 1 channel, if your mp3 is stereo

Then edit your musiconhold.conf like this:

[native]
mode=files
directory=/var/lib/asterisk/moh-native

and you'll have a nice native streaming. You can convert your stuff to
another formats, like "sox file.mp3 [-c1] file.gsm" or "sox file.mp3
[-c1] file.ul" and let asterisk decide which one best fits given
channel.

[]'s
MM

 -Original Message-
From:   "Lee Archer" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Cc: 
Sent:  Sat, 1 Apr 2006 10:34:42 +0100
Delivered:  Sat,  01 Apr 2006 06:28:16
Subject:[Asterisk-Users] 1.2.6 doesn't use mpg123?

I use mpg123 for streaming but I can't get it to compile under SuSe10
and x86_64 CPU.  Does anyone have any recommendations for other programs
that allow streaming and will compile on this arch? 

Regards

Lee

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: 31 March 2006 22:36
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 1.2.6 doesn't use mpg123?

> >
> And isn't mpg123 ( or some replacement ) required when using a stream 
> for MOH I couldn't get streaming to work without it in 1.2?

Yes.. mpg123 is required for streaming... I had it working in 1.0.9...
though have not tried in 1.2.
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Re: Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?

Ok this is great... but I just noticed this morning while doing some
tests that asterisk seems to start a new stream for every caller 
With mpg123 it would just start one and all calls would hear the same
stream.Unless something was seriously lagging, my test calls this
morning all were in different spots in the hold music.   Isn't this
less efficient?


On 4/1/06, Melcon Moraes <[EMAIL PROTECTED]> wrote:
> You don't have to use it in newer versions. Get your mp3, ant convert to
> slin format with sox.
>
> Ex: sox -V file.mp3 [-c1] file.slin
>
> -V: just to show you what's going on
> -c1: convert to 1 channel, if your mp3 is stereo
>
> Then edit your musiconhold.conf like this:
>
> [native]
> mode=files
> directory=/var/lib/asterisk/moh-native
>
> and you'll have a nice native streaming. You can convert your stuff to
> another formats, like "sox file.mp3 [-c1] file.gsm" or "sox file.mp3
> [-c1] file.ul" and let asterisk decide which one best fits given channel.
>
> []'s
> MM
>
>  -Original Message-
> From:   "Lee Archer" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Cc:
> Sent:  Sat, 1 Apr 2006 10:34:42 +0100
> Delivered:  Sat,  01 Apr 2006 06:28:16
> Subject:[Asterisk-Users] 1.2.6 doesn't use mpg123?
>
> I use mpg123 for streaming but I can't get it to compile under SuSe10
> and x86_64 CPU.  Does anyone have any recommendations for other programs
> that allow streaming and will compile on this arch?
>
> Regards
>
> Lee
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> Sent: 31 March 2006 22:36
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
> Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] 1.2.6 doesn't use mpg123?
>
> > >
> > And isn't mpg123 ( or some replacement ) required when using a stream
> > for MOH I couldn't get streaming to work without it in 1.2?
>
> Yes.. mpg123 is required for streaming... I had it working in 1.0.9...
> though have not tried in 1.2.
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>
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>
> --
> Melcon Moraes <[EMAIL PROTECTED]>
>
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Re: [Asterisk-Users] Re: How is Teliax ?


I would not ride on a tracert too much. We use Teliax also and our ISP that
we have at the data center switched there backbones around the same time
Teliax where doing there upgrades. 


For those that have not analyzed how tracert actually works, you can't 
depend on its output to give you factual evidence as to where delays 
occur in an end-to-end path. Each step through the tracert process does 
nothing more then issue an icmp echo request, measuring the response 
time and displaying it. When a high value is reported (eg, at router #10 
for example), there is no way to know (factually) whether that high 
response was from that particular device (#10) or one of the routers 
prior to that address (eg, #3, #5, or #9). All you really know is that 
at the time the icmp was sent, the response was delayed for some reason, 
and it could have been any of the devices prior to the specific one that 
 you thought was the issue.



We started seeing some call issues and when we did a tracert we started
getting some dropped tracert responses on our ISP new backbone(Time Warner)
when our ISP investigated it Time Warner responded that tracerts get a VERY
low priority on their routers and that is why we where seeing these drops. 


The "low priority" comment is one that was started by Cisco folks many 
years ago when router processors were taxed much heavier then current 
day products. Back then, Cisco IOS firmware prioritized various events 
and icmp's were (and still are) low priority events. However, since then 
the processor speeds have significantly increased and off-loading of 
many routing events to card-level processors have occurred. The 
processing of icmp traffic is seldom (if ever) impacted in any 
measurable way in products manufactured in the last five to ten years.



Once Teliax did whatever their last change was fixed all of our issue and we
have not see any call issue in the last 2-3 weeks. Still see drops on the
traces. I would look more at the latency and for dropped packets if you do a
continues ping with setting the size to something other then default and see
if you get any dropped packets or high latency.


Right on!

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Re: [Asterisk-Users] Building Asterisk embedded device

2006-04-01 Thread sam


Jim Houser wrote:


http://gumstix.com/waysmalls.html

 


Thanks for your link. how to build asterisk into this hardware?

Thanks
Sam


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of sam
Sent: Friday, March 31, 2006 8:01 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Building Asterisk embedded device

Hi,

I want to build a PBX base on Asterisk using an embedded device.
Can anyone please recommend an embedded device I can use for doing so?
I will install linux or freebsd in the device.

Thanks
A
_



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Re[2]: [Asterisk-Users] 1.2.6 doesn't use mpg123?

2006-04-01 Thread Melcon Moraes

You don't have to use it in newer versions. Get your mp3, ant convert to
slin format with sox. 

Ex: sox -V file.mp3 [-c1] file.slin

-V: just to show you what's going on
-c1: convert to 1 channel, if your mp3 is stereo

Then edit your musiconhold.conf like this:

[native]
mode=files
directory=/var/lib/asterisk/moh-native

and you'll have a nice native streaming. You can convert your stuff to
another formats, like "sox file.mp3 [-c1] file.gsm" or "sox file.mp3
[-c1] file.ul" and let asterisk decide which one best fits given channel.

[]'s
MM

 -Original Message-
From:   "Lee Archer" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Cc: 
Sent:  Sat, 1 Apr 2006 10:34:42 +0100
Delivered:  Sat,  01 Apr 2006 06:28:16 
Subject:[Asterisk-Users] 1.2.6 doesn't use mpg123?

I use mpg123 for streaming but I can't get it to compile under SuSe10
and x86_64 CPU.  Does anyone have any recommendations for other programs
that allow streaming and will compile on this arch? 

Regards

Lee

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: 31 March 2006 22:36
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 1.2.6 doesn't use mpg123?

> >
> And isn't mpg123 ( or some replacement ) required when using a stream 
> for MOH I couldn't get streaming to work without it in 1.2?

Yes.. mpg123 is required for streaming... I had it working in 1.0.9...
though have not tried in 1.2.
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[Asterisk-Users] Asterisk box with unreliable ping/latency

2006-04-01 Thread Bjorn O

Hello!

Over the last couple of days I’ve been trouble
shooting a strange problem with Asterisk.

First of all, I should say that I’ve been running
Asterisk on a Fedora Core 3 box since last May, but decided to do a
reinstallation of everything due to some problems we’ve had with echos
during conversations (100% SIP based, so no ZAP echos). We are talking about a
low-volume installation running off an Intel Celeron 2.53 GHz box, 512 MB RAM
and some no-name MB (https://www.legendmemory.com/modules.php?op=modload&name=News&file=article&sid=412).
The internet connection is DSL based, no NAT, eight IP addresses.

The problems continued. I did some tests on the DSL line,
which itself seemed fine. But when I connected Asterisk to the network,
suddenly ping became unstable. We are talking about ping varying between 20 ms
and 600 ms over a connection that usually should result in 20 ms ping. So I
decided to eliminate the chance of increased latency due to traffic, bypassed
the switch and hooked the Asterisk box up directly to the router. I even
disabled Asterisk itself. Still, the problem persisted. It’s worth
mentioning that the latency to the router itself was fine all the time, we’re
just referring to the latency of the Asterisk box.

So I started installing [EMAIL PROTECTED] on a spare computer,
about the same specs, only a different motherboard (http://www.dealtime.com/xPF-Asrock_MB_PM800_ASROCK_P4VM800_RTL_P4VM800).
Same issues. At this very moment, I was left with a suspicion that there might
be something with the DSL network, so I contacted the DSL provider and had them
run tests on the line, which all came out just fine. I switched DSL modem/router,
still same problems.

I then decided to do installations of different Linux
distros (no Asterisk installations) on the spare computer, and did some
interesting observings:

- Debian: Better, only occasional (a few percentage of the
pings performed were above normal, and then only 200 ms above normal, compared
to Centos/Asterisk where 30-40% were above normal). Still not 100%

- Mandriva: Perfect latency all the way

- Centos (base installation): Problems equal to the ones
described earlier

I did some further testing, and on a P4 3,0 GHz, 512 MB RAM
and a motherboard which I don’t know the maker /chipet, the CentOS
installation came out with just a small percentage lag in latency.

I have checked for IRQ errors, and there seems to be no
conflicts. I do see the network sharing IRQ with the USB bus, but this is
common on motherboards with everything integrated.

I would assume this can be defined as off topic, as it
clearly does not relate to Asterisk in particular. However, I am writing this
in the hope of that someone might have had similar issues before and possibly
been able to solve the problem. Google has not given me much on the subject. It’s
not that I don’t mind putting together a new box with separate and higher
level of quality components, but I’d much rather learn something from this
experience than giving up ;)

Best regards,

Bjorn

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Re: [Asterisk-Users] Panasonic KXTD 1232 6

2006-04-01 Thread Krzysztof Drewicz


charles napisaĹ�(a):


I want to replace a Telebutler software


It's Telebutler software some simple IVR/CC solution?

auto attendent system that used a 4 port Dialogic board connected to a 
Panasonic KXTD 1232 6 line system. We have spare computers here. How 
do I connect asterisk to this Panasonic system?



IIRC KXTD 1232 is a 12 public, 32 inside access PBX (up to 24 system and
8 pots for faxes or sth like that).
It uses 6 BRI to access PTSN lines so you have to use some multi-port
BRI (active and powered) card.
http://www.eicon.com/worldwide/products/MediaGateways/diva-server-v4bri.htm
or
http://www.beronet.com/index.php?option=com_content&task=view&id=40&Itemid=28&lang=en
The latter costs:
BN8S0 8 Port S0 Card (TE/NT) + - Power Bundles BN Power
984,84EUR [incl. VAT] 849,00EUR [excl.VAT]

This is not very cost-effective solution, and in most cases you have to
use chan_misdn or other not-so-very-popular channel driver.

Why you use plain PBX when you could do as simple as: one two/four span
E1/T1 card, one port connected to Telco, second (3rd,4rd) connected to
channel bank ?


Btw: you could have a 4 times E1 in one PCI card (brand new) for less 
than a 650 USD and some Zhone CB from ebay around 120-150 USD (not very 
much used, i'm using it by my self).



--
Krzysztof Drewicz

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Re: [Asterisk-Users] Echo cancellation problem

2006-04-01 Thread Sergio Chersovani


Avi Miller ha scritto:

Giuseppe wrote:
Can anybody tell me if there is some error or something missing in 
this configuration please?


I have the same card in a few of my servers and the echo canceller 
works just fine. I'm not 100% sure, but something does jump out at me:
Mar 31 16:40:21 WARNING[29878]: chan_capi.c:3334 
show_capi_conf_error: ISDN3: conf_error 0x300b PLCI=0x103

I guess you have to set the old echo facility number in your capi.conf

Sergio
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[Asterisk-Users] Newbie question - sip.conf incoming contexts

2006-04-01 Thread Steve Gladden

Hello!


I've been struggling with the documentation for months on this simple
subject...
I still have not been able to get this concept down...


I have 3 sip accounts (PSTN DID's) that come into my asterisk box
and give me phone service from my itsp via SIP.
I for the life of me have not been able to figure out how to get them to
come in to 3 seperate contexts!

This must be simple but I am missing the point.
All 3 accounts need a register line (I think) in order to work.

The register lines work great but I have not been able to figure out
how to get the other two lines to come into another seperate inbound
context that I have defined other than the one that is specified
in the [general] section of sip.conf

The /extension number does not do the trick for me

I wuld like for these incoming lines (from the same itsp) to truly
land in one of 3 seperate starting contexts in my dialplan based
on what phone number (account) they are.

Thank you very much for your help... this must be simple
but I have not really figured it out in several months of playing
around and reading

I've figured out a TON of other complex things, but this simple
incoming context thing has me a bit stumped.

I've tried a few things in my sip peer like
register=yes which was suggested on a web site but it does not work.

I also tried maing the peer name match the account (phone number)
of the sip account and that did not do it either.

The peers work fine as outgoing but I've not figured out how to make them
work for incoming as my sip itsp requires that I 'register' for inbound
calls.


Steve



















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[Asterisk-Users] INX (Internationalnumber.com)

2006-04-01 Thread voipman

Unable to make outgoing calls from my asterisk using INX (international.com) but incoming works fine. FYI! my asterisk is working fine for Vbuzzer.com
 for incoming and outgoing calls. Please reply with extension.conf and sip.conf section related to INX, if anyone out there using it successfully.
 
Bunch of thanks!
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Re: [Asterisk-Users] How to check if a phone / line is used?

2006-04-01 Thread Ronald Wiplinger


Jerry Jones wrote:

Show channels?


Yes, on Linux


bye

Ronald Wiplinger



On Mar 31, 2006, at 2:09 AM, Ronald Wiplinger wrote:

In the past I used SetGroup and CheckGroup to figure out if my 
allowed providers lines are all used or not.
Since most of my provider have given me a single line anyway, I 
wonder if there is a way to check if this (provider) line is taken 
already.


How can I do that?

Same is with the phone. How can I see in CLI if a phone is now in use 
or not?
"Sip show peers" shows me just if it is on-line, but not if it is in 
a call or not.
In the dialplan I could dial the number and if it is busy, it would 
go to the Voicemail for unavailable or busy. I expect that there is 
just a test function as well, without trying to call.



bye

Ronald Wiplinger


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Re: [Asterisk-Users] kernel recompilation on a asterisk server

2006-04-01 Thread nik600

ok now it works!

thanks
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RE: [Asterisk-Users] kernel recompilation on a asterisk server

2006-04-01 Thread Marco Campos


Is it a 2.6 kernel? Did you included CRC_CCITT and RTC support when
you made the "make menuconfig"?


-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de nik600
Enviada: sábado, 1 de Abril de 2006 10:28
Para: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Assunto: Re: [Asterisk-Users] kernel recompilation on a asterisk server

i've tried

make clean && make && make install in zaptel...but i still get errors...

particularry i get:


wct4xxp: Unknown symbol zt_rbsbits
wct4xxp: Unknown symbol zt_unregister
wct4xxp: Unknown symbol zt_register
wct4xxp: Unknown symbol zt_alarm_notify
zaptel: Unknown symbol crc_ccitt_table
zaptel: Unknown symbol crc_ccitt_table
wct4xxp: Unknown symbol zt_ec_span
wct4xxp: Unknown symbol zt_receive
wct4xxp: Unknown symbol zt_qevent_lock
wct4xxp: Unknown symbol zt_ec_chunk
wct4xxp: Unknown symbol zt_transmit
wct4xxp: Unknown symbol zt_rbsbits
wct4xxp: Unknown symbol zt_unregister
wct4xxp: Unknown symbol zt_register
wct4xxp: Unknown symbol zt_alarm_notify
zaptel: Unknown symbol crc_ccitt_table
zaptel: Unknown symbol crc_ccitt_table
wct4xxp: Unknown symbol zt_ec_span
wct4xxp: Unknown symbol zt_receive
wct4xxp: Unknown symbol zt_qevent_lock
wct4xxp: Unknown symbol zt_ec_chunk
wct4xxp: Unknown symbol zt_transmit
wct4xxp: Unknown symbol zt_rbsbits
wct4xxp: Unknown symbol zt_unregister
wct4xxp: Unknown symbol zt_register
wct4xxp: Unknown symbol zt_alarm_notify
zaptel: Unknown symbol crc_ccitt_table
zaptel: Unknown symbol crc_ccitt_table
wct4xxp: Unknown symbol zt_ec_span
wct4xxp: Unknown symbol zt_receive
wct4xxp: Unknown symbol zt_qevent_lock
wct4xxp: Unknown symbol zt_ec_chunk
wct4xxp: Unknown symbol zt_transmit
wct4xxp: Unknown symbol zt_rbsbits
wct4xxp: Unknown symbol zt_unregister
wct4xxp: Unknown symbol zt_register
wct4xxp: Unknown symbol zt_alarm_notify
zaptel: Unknown symbol crc_ccitt_table
zaptel: Unknown symbol crc_ccitt_table
wct4xxp: Unknown symbol zt_ec_span
wct4xxp: Unknown symbol zt_receive
wct4xxp: Unknown symbol zt_qevent_lock
wct4xxp: Unknown symbol zt_ec_chunk
wct4xxp: Unknown symbol zt_transmit
wct4xxp: Unknown symbol zt_rbsbits
wct4xxp: Unknown symbol zt_unregister
wct4xxp: Unknown symbol zt_register
wct4xxp: Unknown symbol zt_alarm_notify
zaptel: Unknown symbol crc_ccitt_table
zaptel: Unknown symbol crc_ccitt_table
wct4xxp: Unknown symbol zt_ec_span
wct4xxp: Unknown symbol zt_receive
wct4xxp: Unknown symbol zt_qevent_lock
wct4xxp: Unknown symbol zt_ec_chunk
wct4xxp: Unknown symbol zt_transmit
wct4xxp: Unknown symbol zt_rbsbits
wct4xxp: Unknown symbol zt_unregister
wct4xxp: Unknown symbol zt_register
wct4xxp: Unknown symbol zt_alarm_notify
zaptel: Unknown symbol crc_ccitt_table
zaptel: Unknown symbol crc_ccitt_table
wct4xxp: Unknown symbol zt_ec_span
wct4xxp: Unknown symbol zt_receive
wct4xxp: Unknown symbol zt_qevent_lock
wct4xxp: Unknown symbol zt_ec_chunk
wct4xxp: Unknown symbol zt_transmit
wct4xxp: Unknown symbol zt_rbsbits
wct4xxp: Unknown symbol zt_unregister
wct4xxp: Unknown symbol zt_register
wct4xxp: Unknown symbol zt_alarm_notify
zaptel: Unknown symbol crc_ccitt_table
zaptel: Unknown symbol crc_ccitt_table
torisa: Unknown symbol zt_receive
torisa: Unknown symbol zt_ec_chunk
torisa: Unknown symbol zt_lboname
torisa: Unknown symbol zt_transmit
torisa: Unknown symbol zt_rbsbits
torisa: Unknown symbol zt_unregister
torisa: Unknown symbol zt_register
torisa: Unknown symbol zt_alarm_notify

where is the problem?

thanks
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RE: [Asterisk-Users] 1.2.6 doesn't use mpg123?