RE: [Asterisk-Users] Asterisk Indications

[Christopher Lee]

I've had a closer listen to 400*17 through the handpiece rather than just on
speaker phone, and I get the feeling that the Australian ringing tone must
have been tweaked slightly, perhaps with the introduction of the newer
Ericsson AXE exchanges?

400*17 sounds familiar, perhaps the older exchanges (cross-bar?) used that
format?

That said, the 400+420 isn't exactly how my current exchange sounds, but
sounds good to me anyway :-)

I'm looking at tweaking the sounds somewhat more and moving away from the
exchange sounds... I'd actually like to get it sounding more like a Nortel
Meridian system, but I don't yet have any example rings to work off to try
and get it similar sounding.

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Steve Underwood
> Sent: Sunday, 25 January 2004 4:17 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Asterisk Indications
> 
> The correct tone is 400*17 (383 + 417) according to the ITU specs.
> 
> Actually, nothing would use a 17Hz tone - it doesn't pass through a
> 300-3400Hz channel very well :-)
> 
> Regards,
> Steve
> 


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RE: [Asterisk-Users] Warning /:Asterisk.c:255 Listener : Select Returned Error

[Girish Gopinath]

I am also getting this warning. But i noticed that this happens only when i 
issue a shell command from the Asterisk CLI. But it has never affected any 
functionality. My linux box is a very slow one.

Regards...

Girish

From: Frankie Gravato <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: Asterisk <[EMAIL PROTECTED]>
Subject: [Asterisk-Users] Warning /:Asterisk.c:255 Listener : Select 
Returned Error
Date: Sat, 24 Jan 2004 14:49:05 -0500

Hello Asterisk-List Folks

Jan 24 14:39:27 WARNING[1082805040]: asterisk.c:255 listener: Select 
retured error: Interrupted system call
Jan 24 14:39:27 WARNING[1082805040]: asterisk.c:255 listener: Select 
retured error: Interrupted system call
  == Detected 4 licensed G.729 transcoders
Jan 24 14:39:27 WARNING[1074411264]: translate.c:219 calc_cost: Tra

Has  any  one  ever  seen  this before i started getting this recently
after  i  install  my  g729  codecs  is  there something wrong with my
asterisk  that its telling me cause i dont understand whats its asking
me


,



--
Best regards,
Frankie ([EMAIL PROTECTED])
mailto:[EMAIL PROTECTED]
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[Asterisk-Users] Sipura 2000 Transmit Issues? No Sound being passed to caller

[Frankie Gravato]

I've  been  beating  my head for 5 hours to figure out why my asterisk
server or sipura isn't passing my voice over to the caller. It seems i
can  hear  the  caller  but  they  can't  hear  me it seems either the
asterisk or the sipura isn't passing this information.

Here's my setup specs

asterisk  server  0.7.1  - X100P Card - Sipura 2000 - Nufone Service -
Voicepulse Service and DID's

when  i  get  Phone call using the Voicepulse or Pstn the caller can't
hear  me  or  barely  hear me. The Sipura is running Firmware 1.20 and
calls  are  being  passed  using  Ulaw  Codec? Anyone out there in the
asterisk community please oh please help me before i do something that
my asterisk server won't like.




  

-- 
Best regards,
Frankie ([EMAIL PROTECTED])  
mailto:[EMAIL PROTECTED]

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Re: [Asterisk-Users] Asterisk Indications

[Steve Underwood]

Christopher Lee wrote:

Hi Steve,

Interesting... I'm not sure! My copy of the original indications.conf had
400+17, and looking at the wiki it's the same there also
http://www.voip-info.org/tiki-index.php?page=Asterisk%20indications%20defaul
t
I tested 400*17 and it made a difference, but I still think 420+400 sounds
much closer to it... if there's any other Australian users who have
customised the tones and want to try it out let me know what you think or
what tones you're using.
 

The correct tone is 400*17 (383 + 417) according to the ITU specs.

Actually, nothing would use a 17Hz tone - it doesn't pass through a 
300-3400Hz channel very well :-)

Regards,
Steve
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Re: [Asterisk-Users] Bluetooth discussions

[Brian West]

I have seen bluetooth do nothing but grow. More and more devices are
getting bluetooth the headsets can be had for 64 or so at buy.com

bkw

On Sun, 25 Jan 2004, Steve Underwood wrote:

> Hi Don,
>
> A large number of GSM phones and PDAs now have bluetooth. It looks
> likely that through 2004 the majority of GSM phones anywhere above entry
> level will have Bluetooth. My guess is that this will collapse in 2005,
> and bluetooth will be dead soon after. In the meantime, I don't seem
> many people using the Bluetooth feature they have. The headsets cost
> $100 up, and their battery life is very poor. These have been selling
> for 3 years now, so we are not just seeing the typical high cost of a
> new product. The first generation headsets were over $300. Phone to
> notebook and phone to PDA connectivity seems to be the main use right
> now. However, all those products will soon have 802.11 so the Bluetooth
> feature will be superfluous.
>
> Bluetooth just doesn't do anything very well. Other methods of doing
> cordless headset to phone connectivity offer far better battery life,
> and lower cost. 802.11 is now pretty cheap, much faster, more flexible,
> and it going into everything soon.
>
> Bluetooth prescence sounds like a dead end to me. However, if can work
> for an 802.11 unit it probably has a real future. Anyone with 802.11 in
> their phone can we tracked rather well as they move around.
>
> Disclaimer: The value of predictions can go down as well as up :-)
>
> Regards,
> Steve
>
>
> Don Feuer wrote:
>
> > I have seen a number of phones being made by companies in Korea, but do not
> >know much about what has happened to them.  I have seen a host of cellular
> >to Bluetooth phones at COMDEX two years ago, and I am a strong proponent of
> >Bluetooth.
> >
> >It would be good if someone is from Korea or can read Korean to make a list
> >of companies that manufacture these phones and share them with the
> >community.
> >
> >
>
>
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RE: [Asterisk-Users] Asterisk Indications

[Christopher Lee]

Hi Steve,

Interesting... I'm not sure! My copy of the original indications.conf had
400+17, and looking at the wiki it's the same there also
http://www.voip-info.org/tiki-index.php?page=Asterisk%20indications%20defaul
t

I tested 400*17 and it made a difference, but I still think 420+400 sounds
much closer to it... if there's any other Australian users who have
customised the tones and want to try it out let me know what you think or
what tones you're using.

I have just been thinking perhaps the main advantage of letting the SIP
device generate it's own indications is lower bandwidth use, for my setup
this isn't really an issue, and if I can figure out how to directly modify
the tones in the Cisco 7940 I'll have a go at it.

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Stephen Davies
> Sent: Sunday, 25 January 2004 3:34 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Asterisk Indications
> 
> 
> 
> On Sun, 25 Jan 2004, Christopher Lee wrote:
> 
> > The original indications has 400+17/400, but I find that sounds more
> like
> > two beeps (which could possibly be confused with the Australian
> > congestion/busy tones).
> 
> Shouldn't it be 400*17?
> 
> Steve
> 
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FW: [Asterisk-Users] one way choppy sound problem !

[Asterisk]

Hello list,

I've been experiencing choppy sound as well.

The version on Asterisk I was using originally was dated 10/24/03 (I
think), the problem appeared after I updated from that version.

My setup is a little different though. I'm having choppy sound only on
some incoming calls -- from PSTN->PBX (between spans on a TE410) and
PSTN->SIP.

We use Cisco 7940 handsets and we also bridge calls to our existing PBX.

We don't have any trouble with outgoing calls. If a caller tells us we
"sound like we're on a mobile phone that's breaking up", we call them
back and everything is fine.

I've kept rebuilding libpri, zaptel and asterisk on a regular basis, but
it hasn't gone away.

We do have an IRQ problem, I saw someone mention that, so I'll take a
harder look at that problem and see if it solves it.

I though I'd throw this in, because we notice the choppiness when
bridging PSTN->PBX, not just PSTN->SIP.

Regards,
Ben

Benjamin Wakefield
[EMAIL PROTECTED]
http://www.dcsi.net.au/
DCSI - We do Internet.
64 Queen Street
Warragul, VIC 3820 AU
Ph: (+61) 1300 665 575
Fx: (+61) 1300 556 595

-BEGIN GEEK CODE BLOCK-
Version: 3.12
G! d- s: a-- C+ UL++ P+ L++ E W+ N+ o- K- w+$ O--- M-- V? PS !PE Y--
PGP- t 5 X+ R- tv b- DI-- D--- G-- e* h* r- z++
--END GEEK CODE BLOCK--



-Original Message-
From: Dawid Mielnik [mailto:[EMAIL PROTECTED] 
Posted At: Friday, 2 January 2004 8:24 PM
Posted To: Asterisk
Conversation: [Asterisk-Users] one way choppy sound problem !
Subject: [Asterisk-Users] one way choppy sound problem !


Hi all,

I have my asterisk setup as following:

IP   2 x E1
x-lite <---> Asterisk ---> PSTN


When I place a call from x-lite to PSTN, the quality of the sound in the
direction x-lite -> PSTN is very bad. That is, the voice of the x-lite
user,
heard by the PSTN user is choppy and makes communication not very
pleasant.
The sound is choppy as if bits of data were lost. The strange thing is
that
the x-lite user hears the PSTN user fine !

In x-lite, I have swithed off sience detection (transmit silence - yes),
this has improved the sound quality but did not eliminated the problem.
I
have fed a countinious sound into the microphone and still got chops in
the
sound. I have also tried changing the codecs gsm, alaw, ulaw - but I get
the
same problem with all of them. Maybe the problem lies somewhere in audio
buffering settings on x-lite ?

Has anyone ever had this sort of problem and managed to deal with it ? I
would greatly appreciate your help !

Best regards,

Dave


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RE: [Asterisk-Users] Asterisk Indications

[Stephen Davies]

On Sun, 25 Jan 2004, Christopher Lee wrote:

> The original indications has 400+17/400, but I find that sounds more like
> two beeps (which could possibly be confused with the Australian
> congestion/busy tones).

Shouldn't it be 400*17?

Steve

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RE: [Asterisk-Users] Asterisk Indications

[Christopher Lee]

For the benefit of anyone with the same questions or
searching the archives, I’ve solved my problem to the below.

 

The Cisco 7940 (and other SIP devices) generate their own
indication tones of ring etc., I found by placing an Answer before a dial, then
Asterisk will answer the call and be able to provide indications.conf tones
down the line.

 

Eg:

 

exten => 931,1,Answer

exten => 931,2,Dial(SIP/931,20)

exten => 931,3,Voicemail(u931)

exten => 931,102,Voicemail(b931)

exten => 931,103,Hangup

 

Also, I found the ringing tone for Australia included
in indications.conf doesn’t sound quite like I expected… I’ve
done a little toying with it, and found the following sounds a little bit
closer to what you’d expect, but it’s still not quite right:

 

for [au] context in indications.conf

==

ring = 400+420/400,0/200,400+420/400,0/2000

 

The original indications has 400+17/400, but I find that
sounds more like two beeps (which could possibly be confused with the
Australian congestion/busy tones).

 











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christopher Lee
Sent: Sunday, 18 January 2004 2:04
PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk
Indications



 

Hi,

 

Just wondering if someone could better explain how the indications.conf
file actually affects Asterisk?

 

I am using a Cisco 7940 from my Asterisk system, and have set in
indications.conf “country=au” thinking that this would make the
dialtones/call progress sound like the familiar Australian tones?

 

However when I call another extension on my system, it still sounds
like the American ring tone. Does the indications perhaps only effect Analog
FXS cards and not SIP phones?

 

Also, when loading the Asterisk configs as shown below, it displays a
message about “Removed default indication country ‘au’ and at
the end proceeds to set default indication country to ‘au’…
the Removed part has me thinking it’s forgotten all about the particular
indications for au?

 

==cut from Asterisk console===

    -- Unregistered indication country 'us'

Jan 18 14:02:36 NOTICE[262161]: indications.c:390
ast_unregister_indication_coun

try: Removed default indication country 'au'

    -- Unregistered indication country 'au'

    -- Unregistered indication country 'fr'

    -- Unregistered indication country 'de'

    -- Unregistered indication country 'nl'

    -- Unregistered indication country 'uk'

    -- Unregistered indication country 'fi'

    -- Unregistered indication country 'no'

  == Parsing '/etc/asterisk/indications.conf':   ==
Parsing '/etc/asterisk/indic

ations.conf': Found

    -- Registered indication country 'us'

    -- Registered indication country 'au'

    -- Registered indication country 'fr'

    -- Registered indication country 'de'

    -- Registered indication country 'nl'

    -- Registered indication country 'uk'

    -- Registered indication country 'fi'

    -- Registered indication country 'no'

    -- Setting default indication country to 'au'

==

 

Thanks,

Chris Lee

 

Re: [Asterisk-Users] Bluetooth discussions

[Steve Underwood]

Hi Don,

A large number of GSM phones and PDAs now have bluetooth. It looks 
likely that through 2004 the majority of GSM phones anywhere above entry 
level will have Bluetooth. My guess is that this will collapse in 2005, 
and bluetooth will be dead soon after. In the meantime, I don't seem 
many people using the Bluetooth feature they have. The headsets cost 
$100 up, and their battery life is very poor. These have been selling 
for 3 years now, so we are not just seeing the typical high cost of a 
new product. The first generation headsets were over $300. Phone to 
notebook and phone to PDA connectivity seems to be the main use right 
now. However, all those products will soon have 802.11 so the Bluetooth 
feature will be superfluous.

Bluetooth just doesn't do anything very well. Other methods of doing 
cordless headset to phone connectivity offer far better battery life, 
and lower cost. 802.11 is now pretty cheap, much faster, more flexible, 
and it going into everything soon.

Bluetooth prescence sounds like a dead end to me. However, if can work 
for an 802.11 unit it probably has a real future. Anyone with 802.11 in 
their phone can we tracked rather well as they move around.

Disclaimer: The value of predictions can go down as well as up :-)

Regards,
Steve
Don Feuer wrote:

I have seen a number of phones being made by companies in Korea, but do not
know much about what has happened to them.  I have seen a host of cellular
to Bluetooth phones at COMDEX two years ago, and I am a strong proponent of
Bluetooth.
It would be good if someone is from Korea or can read Korean to make a list
of companies that manufacture these phones and share them with the
community.
 



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Re: [Asterisk-Users] billing systems

[Anton]

We have one we are going to start selling in Feb. 
Currently we use it with asterisk, an Excel running ADS and Max TNT voip MVAM 
gatekeeper.

  - Original Message - 
  From: 
  Don Feuer 
  To: [EMAIL PROTECTED] 
  
  Sent: Saturday, January 24, 2004 11:37 
  PM
  Subject: [Asterisk-Users] billing 
  systems
  
  
  Has anyone out 
  there written a good web based convergent billing system out there that they 
  want to sell??
   
  The system needs 
  to work with both voip as well as TDM switches for provisioning and 
  presentment.
   
  If anyone out 
  there knows of a good one that works along with anyone with one please e-mail 
  me at [EMAIL PROTECTED]
   
  Thanks!
   
  Don

[Asterisk-Users] billing systems

[Don Feuer]

Has anyone out there 
written a good web based convergent billing system out there that they want to 
sell??
 
The system needs to 
work with both voip as well as TDM switches for provisioning and 
presentment.
 
If anyone out there 
knows of a good one that works along with anyone with one please e-mail me at [EMAIL PROTECTED]
 
Thanks!
 
Don

[Asterisk-Users] Good Billing Systems

[Don Feuer]

 Does anyone out there have any suggestions on a decent telecommunications
web based billing system for billing and provisioning multiple
applications??

Has anyone written one that they want to sell??

Please e-mail me at [EMAIL PROTECTED]

Thanks!

Don

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Friday, January 23, 2004 6:58 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] RFC3389 support issue with DG104S

At 5:35 PM -0800 1/23/04, Zot O'Connor wrote:
>
>I am getting (with older image):
>
>RFC3389 support incomplete.  Turn off on client if possible
>
>How do I turn that off for the DG104s?  Or if I can't how do I tweak 
>asterisk?
>
>I see posts about ATA-186's having an audiomode, but the closet I came 
>to was inbanddtmf.  I tried =0 and =1, no effect.
>
>Thanks!
>
>--
>Zot O'Connor <[EMAIL PROTECTED]>
>White Knight Hackers, Inc.

Zot -
   Good to see you're getting around to installing * finally!  The error
messages you're seeing are probably unavoidable, as many ATA devices do not
allow the user to turn on/off the VAD comfort noise stuff.  I would suggest
that an ugly method to solve the problem is just to find that debug line in
the source code and comment it out, since you will have no significant
degradation during calls, but you'll see a lot of noisy error messages.

JT
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[Asterisk-Users] Ring/Off-hook Message

[Mindworks Wireless]

Hello,

I am seeing this error:
Jan 24 20:02:53 WARNING[1175660480]: Ring/Off-hook in strange state 6 on
channel 3

on one of my phone lines that comes into my channel bank.  It only happens
on one line, and it still seems to work.  Also, I have had two different
channel banks installed (ADIT 600 & Adtran TA 750) and see the same error
message on incoming phone calls on that line.

I don't believe it is a configuration issue in * as all of my other lines
are in the same group as this line, and it has occurred on 2 different
channel banks.

I saw a post about this occurring on an X100P, but there were no replies.

Thanks,

- Brent

BTW, Current setup is TA 750 to a T100P.

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RE: [Asterisk-Users] Bluetooth discussions

[Don Feuer]

 I have seen a number of phones being made by companies in Korea, but do not
know much about what has happened to them.  I have seen a host of cellular
to Bluetooth phones at COMDEX two years ago, and I am a strong proponent of
Bluetooth.

It would be good if someone is from Korea or can read Korean to make a list
of companies that manufacture these phones and share them with the
community.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Senad Jordanovic
Sent: Saturday, January 24, 2004 9:21 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Bluetooth discussions

Linus Surguy wrote:
>> IRC channel chatter says that there are some new developments with a 
>> cool presence trick that Mark has come up with for bluetooth devices.
>> I know a bit about it, but I think the general population here would 
>> like to see some details if they're available.
> 
> I don't know if this is what you are talking about, but I know of 
> other experiments where a bluetooth enabled server allows bluetooth 
> enabled mobile (cellular) phones to register and then carry two way 
> calls over bluetooth rather than GSM.
> 
> This would be a cool trick if Asterisk could do this too...
> 
> Linus

I have heard of this as well... But apparently a mobile using This service
needs to be CTP (Cordless Telephony Profile) compatible.

At the moment (at last to my knowledge) there is no such mobile phone
available, although Apparently Samsung is to release one soon.

Ta
SJ 

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Re: [Asterisk-Users] 8 lines - best approach

[Tilghman Lesher]

On Friday 23 January 2004 12:18, Paul Mahler wrote:
> On Friday, January 23, 2004 at 8:04 AM, Steven Critchfield wrote:
> > On Fri, 2004-01-23 at 09:30, Darren Martz wrote:
> > > I have 8 lines coming into an existing PBX system and am looking
> > > for a cost
> > > effective way to replace the existing system with Asterisk. We
> > > need some of
> > > the features in Asterisk, including its ability to support remote
> > > offices (long distance savings).
> > >
> > > At first glance this appears to require a T100P card and a channel
> > > bank, but
> > > that seems rather expensive. My estimated price on that would be
> > > roughly $2600 for 8 lines given that system - perhaps my estimate
> > > is way off
> > >
> > > Is there another way that is more cost effective?
> >
> > That number sounds about right. It is likely that it will be less,
> > but budgeting that much for hardware is a good start.
> 
> Do you have to continue to use the existing handsets? You should look
> at replacing the existing phones with SIP phones.

He did say "cost-effective".  Last I checked, 24 SIP phones (unless they
are Grandstreams) will cost far more than a channel bank.

-Tilghman

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Re: [Asterisk-Users] Back to front logging for calls placed through /var/spool/asterisk/outgoing?

[John Todd]

Try this: make your outbound call via a Local channel, and see if 
that gets logged.

Channel: Local/[EMAIL PROTECTED]
MaxRetries: 0
RetryTime: 60
WaitTime: 60
Context: home
Extension: 10
Priority: 1
and then...

[callout]
exten => _X.,1,Dial(Zap/1/${EXTEN})
exten => _X.,2,Congestion
exten => _X.,102,Busy
exten => h,1,Hangup
JT


Here's an example - placing a call to 271536 from local extension 
10.  The call file is:

Channel: Zap/1/271536
MaxRetries: 0
RetryTime: 60
WaitTime: 60
Context: home
Extension: 10
Priority: 1
... and the cdr record generated by * on completion of the call is:

"","","10","home","","Zap/1-1","SIP/cisco-4edb","Dial","sip/cisco||tTr","20
04-01-24 16:02:10","2004-01-24 16:02:13","2004-01-24 
16:02:26",16,13,"ANSWERED","DOCUMENTATION"

"cisco" is the name given in sip.conf for extension 10.  I was 
expecting a cdr entry for the Zap/1 channel.

 Iain







--On Friday, January 23, 2004 21:55:40 -0500 John Todd 
<[EMAIL PROTECTED]> wrote:

Iain -
   Brian I believe is correct, and Kannaiyan perhaps is not correct.
Perhaps you can post the actual values in one of your call spool files so
that we can comment on it more clearly.  Using the "Application:"
statement in an outbound spool file will prevent a CDR from being
created; use "Context:/Extension:/Priority:" methods.  If that fails,
then we have a bug.
JT

At 5:59 PM -0600 1/23/04, Brian West wrote:
NO it will log from a spool file if and only if you ref an extension and
not an application.
bkw

On Fri, 23 Jan 2004, Kannaiyan Natesan wrote:

 There is no CDR for the call from spool outgoing,

 You need to write a patch to solve the same.

 Kannaiyan

 - Original Message -
 From: "Iain Stevenson" <[EMAIL PROTECTED]>
 To: <[EMAIL PROTECTED]>
 Sent: Friday, January 23, 2004 8:27 PM
 Subject: [Asterisk-Users] Back to front logging for calls placed
 through /var/spool/asterisk/outgoing?
 >
 > I've just noticed that if you start a call by writing a file to
 > /var/spool/asterisk/outgoing the cdr created on termination logs the
 > call placed to the local extension - not to the destination in the
 > PSTN.  Hence there is no record of the PSTN number dialled.  I guess
 > most people want
 to
 > log the outgoing portion not the local call leg?  Anyone know of a
 > setting that changes this?
 >
 > >   Iain
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[Asterisk-Users] Re: SayDigits

[Doug Meredith]

"Chris Wilson" <[EMAIL PROTECTED]> wrote:

>Has anyone had this problem:
>
>(When calling to ext. 1010)
>
>Jan 24 10:50:27 WARNING[-1252262992]: file.c:446 ast_openstream: File digits/" does 
>not exist in any format
>Jan 24 10:50:27 WARNING[-1252262992]: file.c:734 ast_streamfile: Unable to open 
>digits/" (format ULAW): No such file or directory
>
><< in Extensions.conf >>
>exten => 1010,1,SayDigits(${CALLERID})
>
>
>/var/lib/asterisk/sounds/digits exists, and there are many files in there. Any idea's?

File permissions?

Doug
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506-854-7997
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[Asterisk-Users] Re: retrans_pkt: Maximum retries exceeded on call

[Doug Meredith]

"Chris Wilson" <[EMAIL PROTECTED]> wrote:

>Hey,
>
>I'm getting an odd message in my logs, and have'nt been able to find much information 
>on it:
>
>Jan 24 00:22:39 WARNING[-1137431632]: chan_sip.c:486 retrans_pkt: Maximum retries 
>exceeded on call [EMAIL PROTECTED] for seqno 102 (Request)

Just guessing here, but it sounds like Asterisk sent a request, didn't
get a reply, sent again, didn't get a reply, and so on until it hit an
internal limit.  If my guess is correct, I suppose there could be many
causes, including:

* Target host down
* No path to the target
* Firewall blocking traffic
* Target host not running SIP, at least on the targeted port.

Doug
-- 
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SystemGuard - Oracle remote support
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506-854-7997
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[Asterisk-Users] Re: disable transfer on outgoing calls?

[tad]

btw, i think i've found an (ugly) work-around. by dumping both callers
into a meetme conference, and only giving one of them the ability to drop
out with #, the other is able to press pound w/out exiting... as long as they
remember who's who, it should be possible to have one caller use # to call
out to a third party, while the other uses # to navigate external
menus.

obviously not an ideal solution, but it might be ok in the short term.

cheers,
tad

On Sat, 24 Jan 2004, tad wrote:

> hi philipp (et al). thanks for the suggestion - but i
> can't seem to get it to work.
>
> i am able to use a local channel to pass arguments to a dial application.
> for example, this successfully enables transfer on one leg of the call:
>
> [incoming]
> exten => s,1, Answer
> exten => s,2, Dial(Local/[EMAIL PROTECTED]/n|15|t)
>
> [outgoing]
> exten => s,1, Dial(Zap/2/9555|15|r)
>
> however, if i create the Local channel from either the manager interface
> or a .call file, a subsequent Dial() call has transfer enabled on both legs, 
> regardless of what
> arguments i pass.
>
> for example, this successfully parks a call, initializes an outbound
> call, and bridges the two - but transfer (and ringthrough) is always
> enabled on both legs:
>
> [incoming]
> exten => s,1, Answer
> exten => s,2, agi,InitCall.agi ;script creates .call file to
> Local/[EMAIL PROTECTED]/n and return context localleg,s,1
> exten => s,3, ParkAndAnnounce(PARKED|60|Console|incoming,i,1)
>
> [outboundleg]
> exten => s,1, Dial(Zap/2/9555|15|r)
>
> [localleg]
> exten => s,1, Wait(2)
> exten => s,2, ParkedCall(701)
>
> in the Console, i see:
> Executing Dial("Local/[EMAIL PROTECTED],2","Zap/2/9555|15|r") in new
> stack
>
> so, either i'm doing something wrong (quite possible) or it looks like
> there are default permissions that are granted to calls that originate in .call 
> files that
> cannot be disabled by Dial?  does anyone out there know anything about
> this - either where it might be documented, or which are the relevant
> source files?
>
> btw, i have tried this without '/n' on the local channel, and with
> transfer=no in my zapata.conf file. neither seemd to have any effect.
>
> thanks again for your time,
> tad
>
> > Subject: Re: [Asterisk-Users] disable transfer on outgoing calls?
> > To: [EMAIL PROTECTED]
> > Organization: AEGEE
> > Reply-To: [EMAIL PROTECTED]
> >
> > Hi!
> >
> > > of the above scenario using 'Dial', and confirmed that it works). however,
> > > i can't seem to figure out how to disable transfer for outgoing calls that
> > > are initiated through the Manager interface (or through .call files, for that 
> > > matter).
> >
> > Interesting problem.
> >
> > Try to use a "Local" channel and an extension in your call file. That way
> > you can trigger a Dial() statement from your dialplan for both legs, and
> > there you should have full control over transfer rights... check the wiki
> > for local channels (and pay attention to the /n option).
> >
> > Cheers, Philipp
> >
> >
> >
>
>

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Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)

[Philipp von Klitzing]

Hi!

> > I've concluded that the Netgear router (FVS318) performing the NAT is
> > corrupting the outgoing RTP packets.  Traces confirmed that the BudgeTone
> > is sending them out with a UDP checksum of 0 but the next hop after the
> > Netgear router they are set to a non-zero value (an incorrect one).
> > Asterisk is never even seeing the packets because the kernel is
> > recognizing them as corrupt and dropping them, hence the recvfrom()
> > "Resource temporarily unavailable" errors in rtp.c.
> 
> Here is Netgear's response:
>  Original Message 
> SIP VOIP phones do not work with netgear routers. The router will always
> set a value in the checksum.

For the record: With a BT 101 behind NAT provided by a Netgear WGR614 I 
don't experience that error message.

Philipp


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Re: [Asterisk-Users] retrans_pkt: Maximum retries exceeded on call

[Master Abi]

I think this is related to a device (GS in my case) that has an sip 
entry but you physically removed it and switched it off. Somehow * still 
thinks connected. Comment out the entry and reload or put the device back.

Mark Rizzo wrote:

I have seen similar error which coincided with my GS phone taking a 
call-waiting call while I was on the GS phone.  I got two of the errors 
(101 102 I think) and then the GS phone or Asterisk terminated the call 
I was on (including the call-waiting call that was trying to get through).

 

I chalked this up to missing configuration setup or that GS does not 
support call-waiting but had not researched yet.

 

Mark

 

-Original Message-
*From:* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *On Behalf Of *Chris Wilson
*Sent:* Saturday, January 24, 2004 12:26 AM
*To:* [EMAIL PROTECTED]
*Subject:* [Asterisk-Users] retrans_pkt: Maximum retries exceeded on call

 

Hey,

 

I'm getting an odd message in my logs, and have'nt been able to find 
much information on it:

 

Jan 24 00:22:39 WARNING[-1137431632]: chan_sip.c:486 retrans_pkt: 
Maximum retries exceeded on call 
[EMAIL PROTECTED] 
 for seqno 102 
(Request)

 

I'm running asterisk with a Cisco 7960G

 

If anyone know's why i'd get this.Any help would be appreciated! =] 
Thanks!

 

Chris

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[Asterisk-Users] Re: disable transfer on outgoing calls?

[tad]

hi philipp (et al). thanks for the suggestion - but i
can't seem to get it to work.

i am able to use a local channel to pass arguments to a dial application.
for example, this successfully enables transfer on one leg of the call:

[incoming]
exten => s,1, Answer
exten => s,2, Dial(Local/[EMAIL PROTECTED]/n|15|t)

[outgoing]
exten => s,1, Dial(Zap/2/9555|15|r)

however, if i create the Local channel from either the manager interface
or a .call file, a subsequent Dial() call has transfer enabled on both legs, 
regardless of what
arguments i pass.

for example, this successfully parks a call, initializes an outbound
call, and bridges the two - but transfer (and ringthrough) is always
enabled on both legs:

[incoming]
exten => s,1, Answer
exten => s,2, agi,InitCall.agi ;script creates .call file to
Local/[EMAIL PROTECTED]/n and return context localleg,s,1
exten => s,3, ParkAndAnnounce(PARKED|60|Console|incoming,i,1)

[outboundleg]
exten => s,1, Dial(Zap/2/9555|15|r)

[localleg]
exten => s,1, Wait(2)
exten => s,2, ParkedCall(701)

in the Console, i see:
Executing Dial("Local/[EMAIL PROTECTED],2","Zap/2/9555|15|r") in new
stack

so, either i'm doing something wrong (quite possible) or it looks like
there are default permissions that are granted to calls that originate in .call files 
that
cannot be disabled by Dial?  does anyone out there know anything about
this - either where it might be documented, or which are the relevant
source files?

btw, i have tried this without '/n' on the local channel, and with
transfer=no in my zapata.conf file. neither seemd to have any effect.

thanks again for your time,
tad

> Subject: Re: [Asterisk-Users] disable transfer on outgoing calls?
> To: [EMAIL PROTECTED]
> Organization: AEGEE
> Reply-To: [EMAIL PROTECTED]
>
> Hi!
>
> > of the above scenario using 'Dial', and confirmed that it works). however,
> > i can't seem to figure out how to disable transfer for outgoing calls that
> > are initiated through the Manager interface (or through .call files, for that 
> > matter).
>
> Interesting problem.
>
> Try to use a "Local" channel and an extension in your call file. That way
> you can trigger a Dial() statement from your dialplan for both legs, and
> there you should have full control over transfer rights... check the wiki
> for local channels (and pay attention to the /n option).
>
> Cheers, Philipp
>
>
>

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[Asterisk-Users] Inter-Fone (was Mediatrix 1204 sip experience?)

[Doug Meredith]

"Jess Magnaye" <[EMAIL PROTECTED]> wrote:

>Go for inter-fone products. it can both support sip and h323.

I took a look at their site, and some of the products look quite
interesting.  The prices don't seem too bad either, if I am
interpreting them right.  It is hard to tell if the prices include the
FXS/FXO modules or not.

Interesting (at least to me) observation number 1:  they have a US
address, but their copy appears to have been written by a person with
a first language other than English.

Observation number 2:  I did a quick web search and couldn't find much
of anything about them.  No re-sellers even.  Perhaps they only sell
direct?

Has anybody actually used any of these boxes?  What did you think?

Doug
-- 
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SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
506-854-7997
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Re: [Asterisk-Users] Has Nufone gone belly-up

[Sathya]

Frankie,

Thanks for your response, and BKW too.

I am not 'thrashing' anybody here. This is my experience and I have seen
people posting their experiences (good, bad) with many other voip providers
on this list.

well, I subscribed to NuFone because I've seen your kind of postings on this
list. May be I am one unfortunate folk. Hope I too can soon have a rock
solid service.

Still waiting


Cheers

Sathya


Message: 8
Date: Sat, 24 Jan 2004 14:36:54 -0500
From: Frankie Gravato <[EMAIL PROTECTED]>
Organization: Cfsdigital
To: Sathya <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] Has Nufone gone belly-up
Reply-To: [EMAIL PROTECTED]

Hello Sathya,

Saturday, January 24, 2004, 12:25:43 PM, you wrote:

S> Folks,

S> I've ordered a new account from Nufone last month. Transferred money to
S> Nufone through their paypal account. I had communication with Nufone
sales
S> up until two weeks back. Since then there were no replies to my emails.

S> I am afraid with this kind of unresponsiveness how one would run a
reliable
S> service with this company. Have no bad feeling with Jeremy as the author
of
S> widely used h323 channel, but my concern is about the company NuFone. Lot
of
S> newcomers when asked for IAX termination/Origination we say NuFone. I
just
S> want to record my experience so far, as it would help anyone wanted to
start
S> with this company. I can live with the fact that they do not have any web
S> based interface for customers to do anything with the service as claimed
by
S> the website. But cannot understand taking two weeks to answer a freaking
S> email. (Well in the absence of trouble ticketing system or web based
access
S> to accounts, email is the only way to contact Nufone)

S> I have services running with Iconnect and Voicepulse etc and I was just
S> trying to use Nufone being well recommended in this list.

S> I am not here to tarnish Nufone name but I have no option but to ask the
S> community since there is no response to my emails or there is no
indication
S> of when my service is available. If they have gone belly-up, well I can
then
S> concentrate on some other company and consider my money as a cost of a
bad
S> choice on my part.

S> If I am a very rare case who just had a bad experience with an excellent
S> company ( I wish ), Nufone please fix this ASSAP.

S> Later

S> Sathya


S> ___
S> Asterisk-Users mailing list
S> [EMAIL PROTECTED]
S> http://lists.digium.com/mailman/listinfo/asterisk-users
S> To UNSUBSCRIBE or update options visit:
S>http://lists.digium.com/mailman/listinfo/asterisk-users


Sathya..

Ripping on Nufone here isn't the greatest thing in the world to do.

I  get  emails  from Nufone as soon as i have issue or problem or if i
See  Jermey on irc he usually answers my questions within couple mins.
I've  been  using  nufone since November its been Rocksolid lot better
support from them then the all mighty voicepulse which takes sometimes
months to get anything out of those fools.

Nufone is great dont be trashing them.


--
Best regards,
Frankie   ([EMAIL PROTECTED])
mailto:[EMAIL PROTECTED]


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RE: [Asterisk-Users] Has Nufone gone belly-up

[C. Johnson]

I don't think Sathya is ripping Nufone per-se,
just trying to figure out what is going on. I'm
sure you would be doing the same thing IF you did
not get a reply, and did not know where to reach
him. Maybe Sathya does NOT know about the
chatroom. Lighten up.


And I agree, if there is a problem, let it be
known to the community. Just don't "sweep it under
the rug" as Roy put it..

-cj

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED]
On Behalf Of Roy
> Sent: Saturday, January 24, 2004 3:23 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Has Nufone gone
belly-up
> 
> 
> I want to hear about problems with VOIP vendors.
Sweeping 
> them under the rug isn't going to help.  If its
a valid 
> problem please post it.
> 
> 
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED]
Behalf Of 
> Frankie Gravato
> Sent: Saturday, January 24, 2004 11:37 AM
> To: Sathya
> Subject: Re: [Asterisk-Users] Has Nufone gone
belly-up
> 
> 
> Hello Sathya,
> 
> Saturday, January 24, 2004, 12:25:43 PM, you
wrote:
> 
> S> Folks,
> 
> S> I've ordered a new account from Nufone last
month. 
> Transferred money 
> S> to Nufone through their paypal account. I had
communication with 
> S> Nufone
> sales
> S> up until two weeks back. Since then there
were no replies 
> to my emails.
> 
> S> I am afraid with this kind of
unresponsiveness how one would run a
> reliable
> S> service with this company. Have no bad
feeling with Jeremy as the 
> S> author
> of
> S> widely used h323 channel, but my concern is
about the 
> company NuFone. 
> S> Lot
> of
> S> newcomers when asked for IAX
termination/Origination we 
> say NuFone. I
> just
> S> want to record my experience so far, as it
would help 
> anyone wanted 
> S> to
> start
> S> with this company. I can live with the fact
that they do 
> not have any 
> S> web based interface for customers to do
anything with the 
> service as 
> S> claimed
> by
> S> the website. But cannot understand taking two
weeks to answer a 
> S> freaking email. (Well in the absence of
trouble ticketing 
> system or 
> S> web based
> access
> S> to accounts, email is the only way to contact
Nufone)
> 
> S> I have services running with Iconnect and
Voicepulse etc and I was 
> S> just trying to use Nufone being well
recommended in this list.
> 
> S> I am not here to tarnish Nufone name but I
have no option 
> but to ask 
> S> the community since there is no response to
my emails or 
> there is no
> indication
> S> of when my service is available. If they have
gone 
> belly-up, well I 
> S> can
> then
> S> concentrate on some other company and
consider my money as 
> a cost of 
> S> a
> bad
> S> choice on my part.
> 
> S> If I am a very rare case who just had a bad
experience with an 
> S> excellent company ( I wish ), Nufone please
fix this ASSAP.
> 
> S> Later
> 
> S> Sathya
> 
> 
> S>
___
> S> Asterisk-Users mailing list
> S> [EMAIL PROTECTED]
> S>
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users
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> S>
http://lists.digium.com/mailman/listinfo/asterisk-
users
> 
> 
> Sathya..
> 
> Ripping on Nufone here isn't the greatest thing
in the world to do.
> 
> I  get  emails  from Nufone as soon as i have
issue or 
> problem or if i See  Jermey on irc he usually
answers my 
> questions within couple mins.
> I've  been  using  nufone since November its
been Rocksolid 
> lot better support from them then the all mighty
voicepulse 
> which takes sometimes months to get anything out
of those fools.
> 
> Nufone is great dont be trashing them.
> 
> 
> --
> Best regards,
> Frankie   ([EMAIL PROTECTED])
> mailto:[EMAIL PROTECTED]
> 
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RE: [Asterisk-Users] Has Nufone gone belly-up

[Roy]

I want to hear about problems with VOIP vendors.  Sweeping them under the
rug isn't going to help.  If its a valid problem please post it.




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Frankie
Gravato
Sent: Saturday, January 24, 2004 11:37 AM
To: Sathya
Subject: Re: [Asterisk-Users] Has Nufone gone belly-up


Hello Sathya,

Saturday, January 24, 2004, 12:25:43 PM, you wrote:

S> Folks,

S> I've ordered a new account from Nufone last month. Transferred money to
S> Nufone through their paypal account. I had communication with Nufone
sales
S> up until two weeks back. Since then there were no replies to my emails.

S> I am afraid with this kind of unresponsiveness how one would run a
reliable
S> service with this company. Have no bad feeling with Jeremy as the author
of
S> widely used h323 channel, but my concern is about the company NuFone. Lot
of
S> newcomers when asked for IAX termination/Origination we say NuFone. I
just
S> want to record my experience so far, as it would help anyone wanted to
start
S> with this company. I can live with the fact that they do not have any web
S> based interface for customers to do anything with the service as claimed
by
S> the website. But cannot understand taking two weeks to answer a freaking
S> email. (Well in the absence of trouble ticketing system or web based
access
S> to accounts, email is the only way to contact Nufone)

S> I have services running with Iconnect and Voicepulse etc and I was just
S> trying to use Nufone being well recommended in this list.

S> I am not here to tarnish Nufone name but I have no option but to ask the
S> community since there is no response to my emails or there is no
indication
S> of when my service is available. If they have gone belly-up, well I can
then
S> concentrate on some other company and consider my money as a cost of a
bad
S> choice on my part.

S> If I am a very rare case who just had a bad experience with an excellent
S> company ( I wish ), Nufone please fix this ASSAP.

S> Later

S> Sathya


S> ___
S> Asterisk-Users mailing list
S> [EMAIL PROTECTED]
S> http://lists.digium.com/mailman/listinfo/asterisk-users
S> To UNSUBSCRIBE or update options visit:
S>http://lists.digium.com/mailman/listinfo/asterisk-users


Sathya..

Ripping on Nufone here isn't the greatest thing in the world to do.

I  get  emails  from Nufone as soon as i have issue or problem or if i
See  Jermey on irc he usually answers my questions within couple mins.
I've  been  using  nufone since November its been Rocksolid lot better
support from them then the all mighty voicepulse which takes sometimes
months to get anything out of those fools.

Nufone is great dont be trashing them.


--
Best regards,
Frankie   ([EMAIL PROTECTED])
mailto:[EMAIL PROTECTED]

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[Asterisk-Users] Incoming DID call Voice Problems

[Bisker, Scott (7805)]

Hello All,

I am experiencing some intermittent problems with calls coming inbound on my DID 
trunk.  I have 12 DIDs that come into an Adtran 750. From there T-1 to a port on 
T400P.  The problem is that some calls that come in don't seem to bridge properly.

Heres what happens.

Call comes in on Trunk.
Call Routed to correct Zap Channel.
Phone Rings.
Person Answers phone, but hears nothing but their own echo.
Calling party hears everything fine.

I have MARK2 enabled in Zaptel driver for echo problems on my PRI line.


I can't seem to replicate the problem calling out PRI to the DIDs, or from a cell 
phone.


I can reliably replicate the problem with an offsite customer that calls in.


Any idea what may be causing this?

Thanks in advance.



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Re: [Asterisk-Users] Asterisk RPMS for RH9 + RH7.3

[Greg Boehnlein]

On Sat, 24 Jan 2004, WipeOut wrote:

> Happy Birthday Greg..
> 
> Have a good one.. :)

Hehehe.. Within 20 minutes, I received a simple patch from [EMAIL PROTECTED] 
that will allow the next release to build on Fedora Core 1 ;)
 
> Greg Boehnlein wrote:
> 
> >Hello all,
> > It's my birthday today, so as my present I would like everyone 
> >possible to download and test my updated set of RPMS for Asterisk 0.7.1. 
> >By popular request, I installed and built a set of RPMS for RedHat 9.0, 
> >and in the process fixed a bunch of issues from the initial build. I have 
> >also updated and will be maintaining a page on the Asterisk Wiki located 
> >at: http://www.voip-info.org/tiki-index.php?page=Asterisk RPM
> >
> >Current Release
> >---
> >asterisk-0.7.1-2.i386.rpm
> >libpri-0.5.1-2.i386.rpm
> >zaptel-0.8.0-2.i386.rpm
> >kernel-module-zaptel-0.8.0-2_2.4.20_28.7.i386.rpm
> >
> >RedHat 7.3
> >--
> >ftp://ftp.nacs.net/asterisk/rh73/RPMS/
> >ftp://ftp.nacs.net/asterisk/rh73/SRPMS/
> >
> >RedHat 9.0
> >--
> >ftp://ftp.nacs.net/asterisk/rh9/RPMS/
> >ftp://ftp.nacs.net/asterisk/rh9/SRPMS/
> >
> >Changelog
> >-
> >* Thu Jan 22 2004 Gregory Boehnlein <[EMAIL PROTECTED]>
> >
> >- added %doc macros
> >- added %config macros
> >- updated %install to correct symlink issue
> >- updated patch0 to include changes to Makefile
> >- added /etc/rc.d/init.d/asterisk
> >- added "export LD_ASSUME_KERNEL=2.4.1" for RH9
> >- asterisk.spec now builds cleanly on RH73 and RH9
> >
> >* Wed Jan 21 2004 Gregory J. Boehnlein <[EMAIL PROTECTED]> 
> >
> >- Initial .spec file created. Most likely buggered. Badly needs help.
> >
> >Readme
> >--
> >Asterisk RPMS for RedHat
> >
> >Welcome to the wonderful world of Open Source VoIP!
> >
> >Asterisk is a complete PBX in software. It runs on Linux and 
> >provides all of the features you would expect from a PBX and 
> >more. Asterisk does voice over IP in three protocols, and can 
> >inter operate with almost all standards-based telephony equipment 
> >using relatively inexpensive hardware.
> >
> >The RPMS contained on this FTP site are experimental builds for 
> >RedHat and are not officially supported or sanctioned by Digium. 
> >Therefore, support will not be provided by Digium. Please be 
> >respectful of this fact and use the high quality, extensive 
> >resources provided by the Asterisk community for support. Chances 
> >are that all of the questions you will ask have been answered in 
> >the past. Take the time to research before you ask!
> >
> >While the maintainers of these RPMS have made every effort to 
> >ensure the widest level of compatibility and stability, every 
> >Asterisk installation is different. Therefore, we would caution 
> >you against using these on Production systems without proper 
> >testing.
> >
> >Currently, these RPMS are being maintained by Gregory Boehnlein 
> ><[EMAIL PROTECTED]>. Please help the cause by sending patches, 
> >updates, suggestions and improvements to me in E-mail.
> >
> >Quickstart
> >--
> >Download the appropriate RPMS for your distribution and install 
> >in the following order:
> >
> >rpm -Uvh libpri*
> >rpm -Uvh zaptel*
> >rpm -Uvh kernel-module-zaptel*
> >rpm -Uvh asterisk*
> >
> >Configuration
> >-
> >Installing the RPMS will not yield a working system and to get 
> >anything useful working, you will need to configure Asterisk. 
> >Please review the following PDF file information on getting 
> >started with Asterisk: http://www.digium.com/handbook-draft.pdf
> >
> >The Asterisk Wiki represents the collective knowledge of the 
> >community and is a resource that you will use often. Please help 
> >the cause by reviewing, correcting and updating the information 
> >found at: http://www.voip-info.org/wiki-Asterisk
> >
> >Special Thanks
> >--
> >These RPMS are made possible from a combination of work that I 
> >have done and the excellent work of Tom Moertel 
> >(http://community.moertel.com) for the Zaptel RPM.
> >
> >  
> >
> 
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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Re: [Asterisk-Users] Asterisk RPMS for RH9 + RH7.3

[WipeOut]

Happy Birthday Greg..

Have a good one.. :)

Greg Boehnlein wrote:

Hello all,
	It's my birthday today, so as my present I would like everyone 
possible to download and test my updated set of RPMS for Asterisk 0.7.1. 
By popular request, I installed and built a set of RPMS for RedHat 9.0, 
and in the process fixed a bunch of issues from the initial build. I have 
also updated and will be maintaining a page on the Asterisk Wiki located 
at: http://www.voip-info.org/tiki-index.php?page=Asterisk RPM

Current Release
---
asterisk-0.7.1-2.i386.rpm
libpri-0.5.1-2.i386.rpm
zaptel-0.8.0-2.i386.rpm
kernel-module-zaptel-0.8.0-2_2.4.20_28.7.i386.rpm
RedHat 7.3
--
ftp://ftp.nacs.net/asterisk/rh73/RPMS/
ftp://ftp.nacs.net/asterisk/rh73/SRPMS/
RedHat 9.0
--
ftp://ftp.nacs.net/asterisk/rh9/RPMS/
ftp://ftp.nacs.net/asterisk/rh9/SRPMS/
Changelog
-
* Thu Jan 22 2004 Gregory Boehnlein <[EMAIL PROTECTED]>
- added %doc macros
- added %config macros
- updated %install to correct symlink issue
- updated patch0 to include changes to Makefile
- added /etc/rc.d/init.d/asterisk
- added "export LD_ASSUME_KERNEL=2.4.1" for RH9
- asterisk.spec now builds cleanly on RH73 and RH9
* Wed Jan 21 2004 Gregory J. Boehnlein <[EMAIL PROTECTED]> 

- Initial .spec file created. Most likely buggered. Badly needs help.

Readme
--
Asterisk RPMS for RedHat

Welcome to the wonderful world of Open Source VoIP!
Asterisk is a complete PBX in software. It runs on Linux and 
provides all of the features you would expect from a PBX and 
more. Asterisk does voice over IP in three protocols, and can 
inter operate with almost all standards-based telephony equipment 
using relatively inexpensive hardware.

The RPMS contained on this FTP site are experimental builds for 
RedHat and are not officially supported or sanctioned by Digium. 
Therefore, support will not be provided by Digium. Please be 
respectful of this fact and use the high quality, extensive 
resources provided by the Asterisk community for support. Chances 
are that all of the questions you will ask have been answered in 
the past. Take the time to research before you ask!

While the maintainers of these RPMS have made every effort to 
ensure the widest level of compatibility and stability, every 
Asterisk installation is different. Therefore, we would caution 
you against using these on Production systems without proper 
testing.

Currently, these RPMS are being maintained by Gregory Boehnlein 
<[EMAIL PROTECTED]>. Please help the cause by sending patches, 
updates, suggestions and improvements to me in E-mail.

Quickstart
--
Download the appropriate RPMS for your distribution and install 
in the following order:

rpm -Uvh libpri*
rpm -Uvh zaptel*
rpm -Uvh kernel-module-zaptel*
rpm -Uvh asterisk*
Configuration
-
Installing the RPMS will not yield a working system and to get 
anything useful working, you will need to configure Asterisk. 
Please review the following PDF file information on getting 
started with Asterisk: http://www.digium.com/handbook-draft.pdf

The Asterisk Wiki represents the collective knowledge of the 
community and is a resource that you will use often. Please help 
the cause by reviewing, correcting and updating the information 
found at: http://www.voip-info.org/wiki-Asterisk

Special Thanks
--
These RPMS are made possible from a combination of work that I 
have done and the excellent work of Tom Moertel 
(http://community.moertel.com) for the Zaptel RPM.

 



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[Asterisk-Users] Warning /:Asterisk.c:255 Listener : Select Returned Error

[Frankie Gravato]

Hello Asterisk-List Folks

Jan 24 14:39:27 WARNING[1082805040]: asterisk.c:255 listener: Select retured error: 
Interrupted system call
Jan 24 14:39:27 WARNING[1082805040]: asterisk.c:255 listener: Select retured error: 
Interrupted system call
  == Detected 4 licensed G.729 transcoders
Jan 24 14:39:27 WARNING[1074411264]: translate.c:219 calc_cost: Tra

Has  any  one  ever  seen  this before i started getting this recently
after  i  install  my  g729  codecs  is  there something wrong with my
asterisk  that its telling me cause i dont understand whats its asking
me



,

  

-- 
Best regards,
Frankie ([EMAIL PROTECTED])  
mailto:[EMAIL PROTECTED]

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Re: [Asterisk-Users] Has Nufone gone belly-up

[Frankie Gravato]

Hello Sathya,

Saturday, January 24, 2004, 12:25:43 PM, you wrote:

S> Folks,

S> I've ordered a new account from Nufone last month. Transferred money to
S> Nufone through their paypal account. I had communication with Nufone sales
S> up until two weeks back. Since then there were no replies to my emails.

S> I am afraid with this kind of unresponsiveness how one would run a reliable
S> service with this company. Have no bad feeling with Jeremy as the author of
S> widely used h323 channel, but my concern is about the company NuFone. Lot of
S> newcomers when asked for IAX termination/Origination we say NuFone. I just
S> want to record my experience so far, as it would help anyone wanted to start
S> with this company. I can live with the fact that they do not have any web
S> based interface for customers to do anything with the service as claimed by
S> the website. But cannot understand taking two weeks to answer a freaking
S> email. (Well in the absence of trouble ticketing system or web based access
S> to accounts, email is the only way to contact Nufone)

S> I have services running with Iconnect and Voicepulse etc and I was just
S> trying to use Nufone being well recommended in this list.

S> I am not here to tarnish Nufone name but I have no option but to ask the
S> community since there is no response to my emails or there is no indication
S> of when my service is available. If they have gone belly-up, well I can then
S> concentrate on some other company and consider my money as a cost of a bad
S> choice on my part.

S> If I am a very rare case who just had a bad experience with an excellent
S> company ( I wish ), Nufone please fix this ASSAP.

S> Later

S> Sathya


S> ___
S> Asterisk-Users mailing list
S> [EMAIL PROTECTED]
S> http://lists.digium.com/mailman/listinfo/asterisk-users
S> To UNSUBSCRIBE or update options visit:
S>http://lists.digium.com/mailman/listinfo/asterisk-users


Sathya..

Ripping on Nufone here isn't the greatest thing in the world to do.

I  get  emails  from Nufone as soon as i have issue or problem or if i
See  Jermey on irc he usually answers my questions within couple mins.
I've  been  using  nufone since November its been Rocksolid lot better
support from them then the all mighty voicepulse which takes sometimes
months to get anything out of those fools.

Nufone is great dont be trashing them.


-- 
Best regards,
Frankie   ([EMAIL PROTECTED]) 
mailto:[EMAIL PROTECTED]

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[Asterisk-Users] Asterisk RPMS for RH9 + RH7.3

[Greg Boehnlein]

Hello all,
It's my birthday today, so as my present I would like everyone 
possible to download and test my updated set of RPMS for Asterisk 0.7.1. 
By popular request, I installed and built a set of RPMS for RedHat 9.0, 
and in the process fixed a bunch of issues from the initial build. I have 
also updated and will be maintaining a page on the Asterisk Wiki located 
at: http://www.voip-info.org/tiki-index.php?page=Asterisk RPM

Current Release
---
asterisk-0.7.1-2.i386.rpm
libpri-0.5.1-2.i386.rpm
zaptel-0.8.0-2.i386.rpm
kernel-module-zaptel-0.8.0-2_2.4.20_28.7.i386.rpm

RedHat 7.3
--
ftp://ftp.nacs.net/asterisk/rh73/RPMS/
ftp://ftp.nacs.net/asterisk/rh73/SRPMS/

RedHat 9.0
--
ftp://ftp.nacs.net/asterisk/rh9/RPMS/
ftp://ftp.nacs.net/asterisk/rh9/SRPMS/

Changelog
-
* Thu Jan 22 2004 Gregory Boehnlein <[EMAIL PROTECTED]>

- added %doc macros
- added %config macros
- updated %install to correct symlink issue
- updated patch0 to include changes to Makefile
- added /etc/rc.d/init.d/asterisk
- added "export LD_ASSUME_KERNEL=2.4.1" for RH9
- asterisk.spec now builds cleanly on RH73 and RH9

* Wed Jan 21 2004 Gregory J. Boehnlein <[EMAIL PROTECTED]> 

- Initial .spec file created. Most likely buggered. Badly needs help.

Readme
--
Asterisk RPMS for RedHat

Welcome to the wonderful world of Open Source VoIP!

Asterisk is a complete PBX in software. It runs on Linux and 
provides all of the features you would expect from a PBX and 
more. Asterisk does voice over IP in three protocols, and can 
inter operate with almost all standards-based telephony equipment 
using relatively inexpensive hardware.

The RPMS contained on this FTP site are experimental builds for 
RedHat and are not officially supported or sanctioned by Digium. 
Therefore, support will not be provided by Digium. Please be 
respectful of this fact and use the high quality, extensive 
resources provided by the Asterisk community for support. Chances 
are that all of the questions you will ask have been answered in 
the past. Take the time to research before you ask!

While the maintainers of these RPMS have made every effort to 
ensure the widest level of compatibility and stability, every 
Asterisk installation is different. Therefore, we would caution 
you against using these on Production systems without proper 
testing.

Currently, these RPMS are being maintained by Gregory Boehnlein 
<[EMAIL PROTECTED]>. Please help the cause by sending patches, 
updates, suggestions and improvements to me in E-mail.

Quickstart
--
Download the appropriate RPMS for your distribution and install 
in the following order:

rpm -Uvh libpri*
rpm -Uvh zaptel*
rpm -Uvh kernel-module-zaptel*
rpm -Uvh asterisk*

Configuration
-
Installing the RPMS will not yield a working system and to get 
anything useful working, you will need to configure Asterisk. 
Please review the following PDF file information on getting 
started with Asterisk: http://www.digium.com/handbook-draft.pdf

The Asterisk Wiki represents the collective knowledge of the 
community and is a resource that you will use often. Please help 
the cause by reviewing, correcting and updating the information 
found at: http://www.voip-info.org/wiki-Asterisk

Special Thanks
--
These RPMS are made possible from a combination of work that I 
have done and the excellent work of Tom Moertel 
(http://community.moertel.com) for the Zaptel RPM.

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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Re: [Asterisk-Users] doublehash patch doesn't work in asterisk 0.7.1

[Iain Stevenson]

This is similar to the last version and applies against the current cvs.

cd asterisk
patch -p0 < Parking.patch
Then the double has transfer should be back.

 Iain



--On Friday, January 16, 2004 6:10 pm -0500 mattf <[EMAIL PROTECTED]> 
wrote:

Hello,

I was using the doublehash.patch that Iain Stevenson had created back in
August to change the transfer key from a single hash "#" to a double-hash
"#". It always patches properly, but when I went from CVS 2004-01-12 to
Asterisk 0.7.1 it doesn't seem to work anymore. I've attached the patch to
this email and I use the following command to patch it:
patch -p1 < ./doublehash.patch
Any help would be great. I would like to get this to work with 0.7.1,
because we are dependant upon the doublehash patch working.
Thanks,

MATT---






Parking.patch
Description: Binary data

AW: [Asterisk-Users] Bluetooth discussions

[Sascha Knific]

Hi

I don´t know if this is CTP compatible but it uses Bluetooth:
http://www.olympia-it.de/cdp.htm  (Sorry german only)

Sascha


> -Ursprüngliche Nachricht-
> Von: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] Im Auftrag von Senad Jordanovic
> Gesendet: Samstag, 24. Januar 2004 18:21
> An: [EMAIL PROTECTED]
> Betreff: RE: [Asterisk-Users] Bluetooth discussions
> 
> Linus Surguy wrote:
> >> IRC channel chatter says that there are some new developments with
a
> >> cool presence trick that Mark has come up with for bluetooth
devices.
> >> I know a bit about it, but I think the general population here
would
> >> like to see some details if they're available.
> >
> > I don't know if this is what you are talking about, but I know of
> > other experiments where a bluetooth enabled server allows bluetooth
> > enabled mobile (cellular) phones to register and then carry two way
> > calls over bluetooth rather than GSM.
> >
> > This would be a cool trick if Asterisk could do this too...
> >
> > Linus
> 
> I have heard of this as well... But apparently a mobile using
> This service needs to be CTP (Cordless Telephony Profile) compatible.
> 
> At the moment (at last to my knowledge) there is no such mobile phone
> available, although Apparently Samsung is to release one soon.
> 
> Ta
> SJ



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[Asterisk-Users] SayDigits

[Chris Wilson]

Has anyone had this problem:
 
(When calling to ext. 1010)
 
Jan 24 10:50:27 WARNING[-1252262992]: file.c:446 
ast_openstream: File digits/" does not exist in any formatJan 24 10:50:27 
WARNING[-1252262992]: file.c:734 ast_streamfile: Unable to open digits/" (format 
ULAW): No such file or directory
<< in Extensions.conf >>
exten => 
1010,1,SayDigits(${CALLERID})
 
 
/var/lib/asterisk/sounds/digits exists, and 
there are many files in there. Any idea's?
Thanks! :)
Chris

Re: [Asterisk-Users] UK BT Interface with asterisk?

[Florian Overkamp]

Hi,

Citeren Kannaiyan Natesan <[EMAIL PROTECTED]>:

> > BT broadband voice uses ATA-186s configured as MGCP devices.
> 
> I think asterisk supports MGCP. I want to configure MGCP with asterisk to
> connect to my BT Broadband Voice.
> Do you have any idea relating to that.

Here's my view. AFAIK asterisk can work with MGCP devices (phones or FXO) but 
I don't think asterisk itself can function _as_ a device like that...

Correct me if I'm wrong, though :)

Florian
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Re: [Asterisk-Users] Has Nufone gone belly-up

[Brian West]

Also on a side note... I have noticed emails from Jeremy back to me
getting caught by spamassassin because of his ip addresses on his dynamic
dsl connection.

bkw
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Re: [Asterisk-Users] Has Nufone gone belly-up

[Brian West]

Have you tried to call them?  Your emails could have been caught up in a
spam filer or such I use nufone daily for our 888 service.  I talk to
Jermey daily.  So I dont know what your beef is but your rant has no place
on this mailing list if you are having problems and have spent any time
trying to get someone on the phone you must be doing something wrong.  I
do know about 2 weeks ago GoDaddy screwed up nufone.net's domain and it
was sent off into LALA land for a few days.

So please keep your rants off the list.

bkw

 On Sat, 24 Jan 2004, Sathya wrote:

> Folks,
>
> I've ordered a new account from Nufone last month. Transferred money to
> Nufone through their paypal account. I had communication with Nufone sales
> up until two weeks back. Since then there were no replies to my emails.
>
> I am afraid with this kind of unresponsiveness how one would run a reliable
> service with this company. Have no bad feeling with Jeremy as the author of
> widely used h323 channel, but my concern is about the company NuFone. Lot of
> newcomers when asked for IAX termination/Origination we say NuFone. I just
> want to record my experience so far, as it would help anyone wanted to start
> with this company. I can live with the fact that they do not have any web
> based interface for customers to do anything with the service as claimed by
> the website. But cannot understand taking two weeks to answer a freaking
> email. (Well in the absence of trouble ticketing system or web based access
> to accounts, email is the only way to contact Nufone)
>
> I have services running with Iconnect and Voicepulse etc and I was just
> trying to use Nufone being well recommended in this list.
>
> I am not here to tarnish Nufone name but I have no option but to ask the
> community since there is no response to my emails or there is no indication
> of when my service is available. If they have gone belly-up, well I can then
> concentrate on some other company and consider my money as a cost of a bad
> choice on my part.
>
> If I am a very rare case who just had a bad experience with an excellent
> company ( I wish ), Nufone please fix this ASSAP.
>
> Later
>
> Sathya
>
>
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Re: [Asterisk-Users] Searching the archives - new engine demo

[Kim Hendrikse]

It's NexTrieve for Linux, I wrote it in C. I don't know what a wintel box is
but it sounds windows-like. That won't work for it.

  - Kim Hendrikse

> What technology is it written using?   I have a wintel box set up (Linux
> coming soon) and would be pleased to host it.
> 
> Regards
> 
> Mike
> - Original Message - 
> From: "Girish Gopinath" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Sunday, January 25, 2004 12:07 AM
> Subject: RE: [Asterisk-Users] Searching the archives - new engine demo
> 
> 
> > Hi,
> >
> > Very good! I tried some of the features and they are really good,
> especially
> > the search within selected months. I have been reading this list for the
> > last 4 months. Answers for some(all) of my doubts are there in the mails
> > posted within these months. Normally when googling, I try different
> > combinations to find out these mails. It is ok, but time consuming. Your
> > search engine is just perfect for people like me and i hope someone will
> > host it.
> >
> > Regards...
> >
> > Girish
> >
> > >From: Kim Hendrikse <[EMAIL PROTECTED]>
> > >Reply-To: [EMAIL PROTECTED]
> > >To: [EMAIL PROTECTED]
> > >Subject: [Asterisk-Users] Searching the archives - new engine demo
> > >Date: Sat, 24 Jan 2004 13:21:13 +0100
> > >
> > >Hi,
> > >
> > >I've placed a demo search engine of the asterisk users archive here:
> > >
> > > http://asterisk.nextrieve.com/cgi-bin/asterisk
> > >
> > >I know there are a number of other ways to search this list that have
> been
> > >suggested and one person suggested that another wasn't necessary but this
> > >engine will do some things that the others can't. Specifically you can
> > >search
> > >within selected months and from specific users. Or require a certain word
> > >in a
> > >subject. You also do a fuzzy search if you are not sure of the spelling,
> > >which
> > >approximates a phrase search.
> > >
> > >I'll leave this demo up for a couple of days, if this is interesting to
> > >people
> > >and someone wishes to host it I can provide the code.
> > >
> > >I've noticed myself that it can be difficult to search the list within
> > >certain
> > >time periods. Google simply won't do that. Not base upon the time the
> mail
> > >was
> > >sent.
> > >
> > >Make sure you check out the advance features which is where you can
> > >restrict
> > >to sender, exclude words etc.
> > >
> > >   - Kim Hendrikse
> > >___
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> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > _
> > Rethink your business approach for the new year with the helpful tips
> here.
> > http://special.msn.com/bcentral/prep04.armx
> >
> > ___
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Re: [Asterisk-Users] retrans_pkt: Maximum retries exceeded on call

[Chris Wilson]

Hm, had that enabled since i set everything up. I 
tried with nat=no as well, same problem.
 
Welp, I guess if anyone figgers it out i'd 
appreciate any help that comes my way :).
 
Thanks!
Chris

  - Original Message - 
  From: 
  Kannaiyan 
  Natesan 
  To: [EMAIL PROTECTED] 
  
  Sent: Saturday, January 24, 2004 12:53 
  AM
  Subject: Re: [Asterisk-Users] 
  retrans_pkt: Maximum retries exceeded on call
  
  You are having Cisco 7960G behind NAT.
   
  Try with  nat=yes
   
  I'm not sure any other settings will solve that in 
  asterisk.
  I have tried but no luck.
   
  Kannaiyan
   
  
- Original Message - 
From: 
Chris 
Wilson 
To: [EMAIL PROTECTED] 

Sent: Saturday, January 24, 2004 8:26 
AM
Subject: [Asterisk-Users] retrans_pkt: 
Maximum retries exceeded on call

Hey,
 
I'm getting an odd message in my logs, and 
have'nt been able to find much information on it:
 
Jan 24 00:22:39 WARNING[-1137431632]: 
chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 102 (Request)

I'm running asterisk with a Cisco 7960G 

 
If anyone know's why i'd get this.Any help 
would be appreciated! =] Thanks! 
 
Chris

[Asterisk-Users] Has Nufone gone belly-up

[Sathya]

Folks,

I've ordered a new account from Nufone last month. Transferred money to
Nufone through their paypal account. I had communication with Nufone sales
up until two weeks back. Since then there were no replies to my emails.

I am afraid with this kind of unresponsiveness how one would run a reliable
service with this company. Have no bad feeling with Jeremy as the author of
widely used h323 channel, but my concern is about the company NuFone. Lot of
newcomers when asked for IAX termination/Origination we say NuFone. I just
want to record my experience so far, as it would help anyone wanted to start
with this company. I can live with the fact that they do not have any web
based interface for customers to do anything with the service as claimed by
the website. But cannot understand taking two weeks to answer a freaking
email. (Well in the absence of trouble ticketing system or web based access
to accounts, email is the only way to contact Nufone)

I have services running with Iconnect and Voicepulse etc and I was just
trying to use Nufone being well recommended in this list.

I am not here to tarnish Nufone name but I have no option but to ask the
community since there is no response to my emails or there is no indication
of when my service is available. If they have gone belly-up, well I can then
concentrate on some other company and consider my money as a cost of a bad
choice on my part.

If I am a very rare case who just had a bad experience with an excellent
company ( I wish ), Nufone please fix this ASSAP.

Later

Sathya


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RE: [Asterisk-Users] Bluetooth discussions

[Senad Jordanovic]

Linus Surguy wrote:
>> IRC channel chatter says that there are some new developments with a
>> cool presence trick that Mark has come up with for bluetooth devices.
>> I know a bit about it, but I think the general population here would
>> like to see some details if they're available.
> 
> I don't know if this is what you are talking about, but I know of
> other experiments where a bluetooth enabled server allows bluetooth
> enabled mobile (cellular) phones to register and then carry two way
> calls over bluetooth rather than GSM.  
> 
> This would be a cool trick if Asterisk could do this too...
> 
> Linus

I have heard of this as well... But apparently a mobile using
This service needs to be CTP (Cordless Telephony Profile) compatible.

At the moment (at last to my knowledge) there is no such mobile phone
available, although Apparently Samsung is to release one soon.

Ta
SJ 

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Re: [Asterisk-Users] exten=>h and ResetCDR

[Brian West]

Exten h isn't needed at all to record CDR info.  Also exten h won't run if
you park the call.

bkw

On Sat, 24 Jan 2004, Girish Gopinath wrote:

> Hi friends,
>
> I have the entry exten => h,Hangup in my extensions.conf, and I am trying to
> record the call details for billing. From the wiki i found out that the use
> of "exten=>h,..." is not suggested for the CDRs. What impact will the use of
> 'h' make on CDRs? Also, what is the advantage of using ResetCDR with
> exten=>h?
>
> Regards...
>
> Girish
>
> _
> Easiest Money Transfer to India.  http://go.msnserver.com/IN/41490.asp Send
> Money To 6000 Indian Towns.
>
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[Asterisk-Users] (no subject)

[Lee Edwards]

Re: [Asterisk-Users] Bluetooth discussions

[Linus Surguy]

> IRC channel chatter says that there are some new developments with a
> cool presence trick that Mark has come up with for bluetooth devices.
> I know a bit about it, but I think the general population here would
> like to see some details if they're available.

I don't know if this is what you are talking about, but I know of other
experiments where a bluetooth enabled server allows bluetooth enabled mobile
(cellular) phones to register and then carry two way calls over bluetooth
rather than GSM.

This would be a cool trick if Asterisk could do this too...

Linus


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[Asterisk-Users] SAFE_ASTERISK DIED - EXIT CODE 127

[Jeroen]

Hi,

Wanted to test Asterisk in safemode using safe_asterisk. Tried to add 
the command to the bootup sequence and tried it via the command
line.  but Asterisk refuses to start up in that mode (it died with 
code 127)

   [EMAIL PROTECTED] /usr/sbin/safe_asterisk
   [EMAIL PROTECTED] Asterisk ended with exit status 127
   Asterisk died with code 127.  Aborting.
Saw the problem in another thread as well - so my question is how do you 
start * as daemon when booting the server?
http://lists.digium.com/pipermail/asterisk-cvs/2004-January/000759.html

Thanks
Jeroen
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[Asterisk-Users] Bluetooth discussions

[John Todd]

IRC channel chatter says that there are some new developments with a 
cool presence trick that Mark has come up with for bluetooth devices. 
I know a bit about it, but I think the general population here would 
like to see some details if they're available.

Mark - care to give the list a rundown on the topic?

In addition, are there plans for a bluetooth channel driver?  The 
Jabra headset is pretty nice, though of course it's answer-only. 
Integration into a chan_jabra would be cool, especially if someone 
gets sphinx voice recognition working...

JT
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RE: [Asterisk-Users] retrans_pkt: Maximum retries exceeded on call

[Mark Rizzo]

I have seen similar error which coincided
with my GS phone taking a call-waiting call while I was on the GS phone.  I got
two of the errors (101 102 I think) and then the GS phone or Asterisk
terminated the call I was on (including the call-waiting call that was trying
to get through).

 

I chalked this up to missing configuration
setup or that GS does not support call-waiting but had not researched yet.

 

Mark

 

-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Wilson
Sent: Saturday, January 24, 2004
12:26 AM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users]
retrans_pkt: Maximum retries exceeded on call

 



Hey,





 





I'm getting an odd message in my
logs, and have'nt been able to find much information on it:





 





Jan 24 00:22:39
WARNING[-1137431632]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on
call [EMAIL PROTECTED]
for seqno 102 (Request)





 





I'm running asterisk with a Cisco
7960G 





 





If anyone know's why i'd get
this.Any help would be appreciated! =] Thanks! 





 





Chris

RE: [Asterisk-Users] Re: Grandstream 100 sidetone

[Mark Rizzo]

I notice the same issue, I think the 'sidetone' is so low that you are able
to hear faint far-end echo.  If the sidetone was louder you would never hear
the far-end echo.

I am new to Grand Stream, has anyone directly asked them to help fix/support
this?

Mark
Perpetual Entertainment, INC.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Meredith
Sent: Saturday, January 24, 2004 6:37 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: Grandstream 100 sidetone

dkwok <[EMAIL PROTECTED]> wrote:

>For people who are using GS 101, what do you think the sidetone 
>generated by the phone.

Seems fine on the two we have.

>I find mind a bit annoying. It has a delay and you notice it as an echo. 
>The volume of the sidetone is also quite hight. I am distracted when 
>both caller and called party talking over each other occasssionally.

Interesting, we don't experience this.  Are you sure this is side tone
and not a far-end echo?

Doug
-- 
Doug Meredith ([EMAIL PROTECTED])
SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
506-854-7997
www.systemguard.com

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Re: [Asterisk-Users] looking for iax termination

[Daniel Bichara]

Hi,

We have termination based on IAX and SIP at Brazil.

Daniel

[EMAIL PROTECTED] wrote:

  
  
  
  Hi,
      I am looking for voip
termination all over the world especially based on IAX or SIP.
      
  Regards.




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Re: [Asterisk-Users] Back to front logging for calls placed through /var/spool/asterisk/outgoing?

[Iain Stevenson]

Here's an example - placing a call to 271536 from local extension 10.  The 
call file is:

Channel: Zap/1/271536
MaxRetries: 0
RetryTime: 60
WaitTime: 60
Context: home
Extension: 10
Priority: 1
... and the cdr record generated by * on completion of the call is:

"","","10","home","","Zap/1-1","SIP/cisco-4edb","Dial","sip/cisco||tTr","20
04-01-24 16:02:10","2004-01-24 16:02:13","2004-01-24 
16:02:26",16,13,"ANSWERED","DOCUMENTATION"

"cisco" is the name given in sip.conf for extension 10.  I was expecting a 
cdr entry for the Zap/1 channel.

 Iain







--On Friday, January 23, 2004 21:55:40 -0500 John Todd <[EMAIL PROTECTED]> 
wrote:

Iain -
   Brian I believe is correct, and Kannaiyan perhaps is not correct.
Perhaps you can post the actual values in one of your call spool files so
that we can comment on it more clearly.  Using the "Application:"
statement in an outbound spool file will prevent a CDR from being
created; use "Context:/Extension:/Priority:" methods.  If that fails,
then we have a bug.
JT

At 5:59 PM -0600 1/23/04, Brian West wrote:
NO it will log from a spool file if and only if you ref an extension and
not an application.
bkw

On Fri, 23 Jan 2004, Kannaiyan Natesan wrote:

 There is no CDR for the call from spool outgoing,

 You need to write a patch to solve the same.

 Kannaiyan

 - Original Message -
 From: "Iain Stevenson" <[EMAIL PROTECTED]>
 To: <[EMAIL PROTECTED]>
 Sent: Friday, January 23, 2004 8:27 PM
 Subject: [Asterisk-Users] Back to front logging for calls placed
 through /var/spool/asterisk/outgoing?
 >
 > I've just noticed that if you start a call by writing a file to
 > /var/spool/asterisk/outgoing the cdr created on termination logs the
 > call placed to the local extension - not to the destination in the
 > PSTN.  Hence there is no record of the PSTN number dialled.  I guess
 > most people want
 to
 > log the outgoing portion not the local call leg?  Anyone know of a
 > setting that changes this?
 >
 > >   Iain
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RE: [Asterisk-Users] Is it possible to push the media processing off to a gateway for processing?

[John Todd]

If you are considering such a service, you need to develop a more 
thorough understanding of VoIP protocols and methods for load 
distribution.  To echo what Stephen Critchfield said to someone else 
just a few hours ago: it's not simple, and you'll probably need a 
consultant.  After you've spent some time designing and discussing, 
you'll probably be able to do it yourself next time but to try this 
from scratch is probably a very long trial-and-error effort.

The short answer to your question for your research is: use load 
sharing and SIP redirects to spread load across multiple boxes, and 
yes, asterisk configs can be pulled from mysql in various ways - dig 
through the distribution for details.

JT


Well, I like the features asterisk gives me such as voicemail and IVR,
Prompts, etc.  I would like to offer an IP Centrex like service, but don't
believe that I can handle very large amounts of users on a box.  The reason
I believe this is that the box would be doing all the media processing/DSP
work on the processor and would be bound by the speed and memory of the box
as to how many simultaneous sessions it could manage.  A gateway has DSP's
which are designed to handle this processing.  I know they are more
expensive, but I could handle large amounts of call volume this way and
still keep the features asterisk offers. 

Another question I also meant to ask was having the ability to read
extensions from a database instead of a .conf file.  I was curious if anyone
has asterisk pulling configs from a database like mysql.
Todd

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Friday, January 23, 2004 9:05 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Is it possible to push the media processing
off to a gateway for processing?
I was wondering if it is possible to have Asterisk push the media
processing
off to something with DSP's such as a gateway?  That way, asterisk just has
to handle the call setups and tear downs.
Todd Wallace
You mean, like what SIP does by default?  This is an incomplete
question.  Please be more specific.  If I have a "gateway", and I
have SIP calls coming in from desktop SIP UA's (hardphones or
softphones) then Asterisk can simply re-direct those calls to the
gateway.
Of course, Asterisk _is_ a gateway, so unless you have specific
reasons for doing so, it would make more sense to use Asterisk to
tackle those jobs with generic, cheap processing horsepower rather
than expensive, proprietary DSP's.
If you're just getting Asterisk to handle call setups and teardowns,
why not just use a real SIP proxy for that?  Or do you not know
enough about your question to understand why I would differentiate
between the two?  (not being nasty here, just wondering if I need to
explain more)
JT
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Re: [Asterisk-Users] SIP register/auth with Grandstream BudgeTone-100

[bfracall]

Hi,Key Aavoja,
Have you successfully registed to * with secret specificated?
Regards.

bfrac

- Original Message - 
From: "Key Aavoja" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, January 23, 2004 2:00 AM
Subject: [Asterisk-Users] SIP register/auth with Grandstream BudgeTone-100


> Hello,
>
> I have a problem with asterisk and Grandstream BudgeTone-100.
> With default configuration everything works (in anonymous mode and fixed
> IP), but if Im trying to enable registering, it dos not work.
> I used 'sip debug' and verbose level 10, nothing happens if I switch
> telephone on (no messages about bad auth etc). As I understood, after
> switching phone on at first it will try to register in asterisk if Im
> trying to call somewhere.
>
> I searched in list-archive and I didnt found that anybody else has this
> kind of problem. I read also:
> http://lists.digium.com/pipermail/asterisk-users/2003-June/013288.html
> and I did so.
>
> sip.conf
> -
> [general]
> port = 5060 ; Port to bind to
> bindaddr = 0.0.0.0  ; Address to bind to
> context = default   ; Default for incoming calls
> disallow=all; Disallow all codecs
> allow=g729
>
> [cisco]
> context=in
> type=friend
> insecure=yes
> host=
> dtmfmode=rfc2833
>
> [grandstream1]
> type=friend
> secret=grandstream1
> host=dynamic
> context=class1
> dtmfmode=rfc2833
>
> [grandstream2]
> type=friend
> secret=grandstream2
> nat=yes
> host=dynamic
> context=class1
> dtmfmode=rfc2833
>
> Asterisk ver: Asterisk CVS-01/22/04-18:13:23
>
> Grandstream ver: Program--1.0.3.81Bootloader--1.0.0.7
HTML--1.0.0.18
>
> * And as I mentioned before, without registration and with static IP
> everything works, it seems, that something is misconfigured in my setup
> for authentication or this phone firmware is buggy? (but its latest, I
> checked www.grandstream.com)
>
>
>
> 
> Best Regards:
>Key Aavoja
>
>
>
>
> /* Never argue with an idiot. They drag you down to their level, then beat
> you with experience.*/
>
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[Asterisk-Users] looking for iax termination

[bfracall]

Hi,
    I am looking for voip 
termination all over the world especially based on IAX or SIP.
    
Regards.

Re: [Asterisk-Users] Debian Packages and Mirrors

[Steven Critchfield]

Of course unstable has 0.7.1 already in the main disribution.

On Fri, 2004-01-23 at 13:48, FastJack wrote:
> hi everybody...
> 
> have you checked the asterisk backports from www.backports.org? I'm
> currently building my asterisk system and i think i will use these debs as
> I've successfully used alot of debs from backports.org in almost every
> production-server we have.
> 
> don't know the quality of the asterisk packages from backports.org but I'm
> almost sure they are great ;))
> 
> bye
> thorsten
> 
> - Original Message -
> From: "William Waites" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Friday, January 23, 2004 5:08 PM
> Subject: [Asterisk-Users] Debian Packages and Mirrors
> 
> 
> > FYI and to whom it may concern, I have made Debian
> > packages of Asterisk et. al. You still need to build
> > a new kernel and the zaptel modules from source, but
> > Asterisk and libpri are manageable with dpkg.
> >
> > The debs as well as mirrors of the source distribution
> > are here:
> >
> > http://www.ntgos.com/Projects/Asterisk/Download
> > http://parc.styx.org/asterisk
> >
> > I would also like to mirror the CVS repository as
> > well as set up a cvsweb...
> >
> > -w
> > --
> > /~\  The ASCII Ribbon Campaign
> > \ /No HTML/RTF in email
> >  X No Word docs in email
> > / \  Respect for open standards
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
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-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] RFC3389 support issue with DG104S

[Pavel Litvinenko]

Zot O'Connor wrote:

I am getting (with older image):

RFC3389 support incomplete.  Turn off on client if possible

How do I turn that off for the DG104s?  Or if I can't how do I tweak
asterisk?
I see posts about ATA-186's having an audiomode, but the closet I came
to was inbanddtmf.  I tried =0 and =1, no effect.
Thanks!

 

ggdbg>set coding
set coding
CODING PROFILE RELATED:
 set coding [prof_id] coding_type [tx|rx] 
[g711_mu,g711_a,a16,a24,a32,a40,g729ab,g723_53,g723_63,fax,fax_t38,clear_chan,clear_chan,pktsig]
 set coding [prof_id] usage  

 set coding [prof_id] vif [rx|tx] 
 set coding [prof_id] encap [t3|frf11|rtp|udp|aal2]
 set coding [prof_id] vad [on|off]
 set coding [prof_id] vad_thresh 
...
Turn vad off

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Re: [Asterisk-Users] DG104S firmware has error?

[Pavel Litvinenko]

Zot O'Connor wrote:

I am installing a used DG104S

I got it to ring from gnophone, but all I got was fast busies.  so I
upgraded based on Pavel's link:


ftp://ftp.dlink.ru/pub/VoIP/DG-104S/Firmware/MGCPDG104.zip

So I now have:

   PROM Version: 3.0B22-DRUNTIME Version: 3.0B44-D

But when I pick the phone up I get:
ggdbg>01604 DIM: 0 DSP ERROR: Reason= DIM ERROR: State Timeout 
01604 DIM: 0:*, State Timeout Error (State = WAIT_RESTART_IND)
01604 DIM: 0:*, BRINGING DSP DOWN !!
01604 DSP 0 Failure. Error: -1
01604 CCU: 0, DSP Failure. State: 2
01604 xGCP: ERROR 0X105 in .\src\ggxgcpif.c at line 3761 (coding
err)

01604 CCU: 0, Coding now (officially) free
01604 tcid 0. NO DSP RESOURCES
01604 xgcpimh: CA message rejected
Is there another setting I have that is causing this?  I did a factory
reset.
I then downgraded to
dg104S_runtime_b35.bin
 

upload your old prom version too ...

What hardware version is on your dg104s ?

But it would not keep the changes to the CA string, either via the web
or console...
Any ideas?

Thanks!

 

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[Asterisk-Users] Re: Grandstream 100 sidetone

[Doug Meredith]

dkwok <[EMAIL PROTECTED]> wrote:

>For people who are using GS 101, what do you think the sidetone 
>generated by the phone.

Seems fine on the two we have.

>I find mind a bit annoying. It has a delay and you notice it as an echo. 
>The volume of the sidetone is also quite hight. I am distracted when 
>both caller and called party talking over each other occasssionally.

Interesting, we don't experience this.  Are you sure this is side tone
and not a far-end echo?

Doug
-- 
Doug Meredith ([EMAIL PROTECTED])
SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
506-854-7997
www.systemguard.com

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RE: [Asterisk-Users] MGCP w/8x8 DTA-310 and as5300 pstn gateway

[Jeremy Jones]

Hi,

Yep, I got the latest firmware (and the next-to-latest, and the
next-to-next-to-latest, and one earlier yet) for SIP.  The first three
(firmware versions 1228, 1227, and 1226) all have that password protected
"Advanced Configuration" page.  The fourth one I found (version ) is a
bit more open.

It appears mum's the word on that password, none of the requests others have
made in, say, vonage forums for that top secret password have met with much
success.  I suppose I could try to brute force it, but I'm fairly lazy, &
I'd just as soon wait for someone to just blurt it out on accident in casual
conversation. 

I did poke around in the binary files & found the html stuff, as well as
what appear to be clear-text default options for stuff one would find on the
protected Advanced Configuration page.  It may be possible for me to use one
of the newer firmware versions, changing the options in the firmware binary
before sticking it on the device, but I imagine I'd screw up a checksum
somewhere if I edit the file directly.  Haven't tried yet.

So, next stop:  sip with the  firware.  

Thanks,

Jeremy

>You can get the latest SIP firmware from Packet8's TFTP server at
>4.42.235.170 file name "current".  Read more about it here
>http://web.packet8.net/download/
>
>Only problems is, in this version the advanced configuration page with
>the SIP setup is password protected.  If you look at the downloaded
>file, you can see all the HTML stuff for the configuration pages.  It may
>be possible to figure out or remove the password protection.
>
>The other option is to load an older version of the SIP firmware in which
>the SIP page is not protected.  I'm sure someone has a copy of it.
>
>By the way, do you have a copy of the MGCP firmware in case you
>want to go back to it?


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RE: [Asterisk-Users] Is it possible to push the media processing off to a gateway for processing?

[Todd Wallace]

Well, I like the features asterisk gives me such as voicemail and IVR,
Prompts, etc.  I would like to offer an IP Centrex like service, but don't
believe that I can handle very large amounts of users on a box.  The reason
I believe this is that the box would be doing all the media processing/DSP
work on the processor and would be bound by the speed and memory of the box
as to how many simultaneous sessions it could manage.  A gateway has DSP's
which are designed to handle this processing.  I know they are more
expensive, but I could handle large amounts of call volume this way and
still keep the features asterisk offers.  

Another question I also meant to ask was having the ability to read
extensions from a database instead of a .conf file.  I was curious if anyone
has asterisk pulling configs from a database like mysql.

Todd


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Friday, January 23, 2004 9:05 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Is it possible to push the media processing
off to a gateway for processing?

>I was wondering if it is possible to have Asterisk push the media
processing
>off to something with DSP's such as a gateway?  That way, asterisk just has
>to handle the call setups and tear downs.
>
>Todd Wallace

You mean, like what SIP does by default?  This is an incomplete 
question.  Please be more specific.  If I have a "gateway", and I 
have SIP calls coming in from desktop SIP UA's (hardphones or 
softphones) then Asterisk can simply re-direct those calls to the 
gateway.

Of course, Asterisk _is_ a gateway, so unless you have specific 
reasons for doing so, it would make more sense to use Asterisk to 
tackle those jobs with generic, cheap processing horsepower rather 
than expensive, proprietary DSP's.

If you're just getting Asterisk to handle call setups and teardowns, 
why not just use a real SIP proxy for that?  Or do you not know 
enough about your question to understand why I would differentiate 
between the two?  (not being nasty here, just wondering if I need to 
explain more)

JT
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Re: [Asterisk-Users] Searching the archives - new engine demo

[Mike Nash [Tall Emu]]

What technology is it written using?   I have a wintel box set up (Linux
coming soon) and would be pleased to host it.

Regards

Mike
- Original Message - 
From: "Girish Gopinath" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, January 25, 2004 12:07 AM
Subject: RE: [Asterisk-Users] Searching the archives - new engine demo


> Hi,
>
> Very good! I tried some of the features and they are really good,
especially
> the search within selected months. I have been reading this list for the
> last 4 months. Answers for some(all) of my doubts are there in the mails
> posted within these months. Normally when googling, I try different
> combinations to find out these mails. It is ok, but time consuming. Your
> search engine is just perfect for people like me and i hope someone will
> host it.
>
> Regards...
>
> Girish
>
> >From: Kim Hendrikse <[EMAIL PROTECTED]>
> >Reply-To: [EMAIL PROTECTED]
> >To: [EMAIL PROTECTED]
> >Subject: [Asterisk-Users] Searching the archives - new engine demo
> >Date: Sat, 24 Jan 2004 13:21:13 +0100
> >
> >Hi,
> >
> >I've placed a demo search engine of the asterisk users archive here:
> >
> > http://asterisk.nextrieve.com/cgi-bin/asterisk
> >
> >I know there are a number of other ways to search this list that have
been
> >suggested and one person suggested that another wasn't necessary but this
> >engine will do some things that the others can't. Specifically you can
> >search
> >within selected months and from specific users. Or require a certain word
> >in a
> >subject. You also do a fuzzy search if you are not sure of the spelling,
> >which
> >approximates a phrase search.
> >
> >I'll leave this demo up for a couple of days, if this is interesting to
> >people
> >and someone wishes to host it I can provide the code.
> >
> >I've noticed myself that it can be difficult to search the list within
> >certain
> >time periods. Google simply won't do that. Not base upon the time the
mail
> >was
> >sent.
> >
> >Make sure you check out the advance features which is where you can
> >restrict
> >to sender, exclude words etc.
> >
> >   - Kim Hendrikse
> >___
> >Asterisk-Users mailing list
> >[EMAIL PROTECTED]
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>
> _
> Rethink your business approach for the new year with the helpful tips
here.
> http://special.msn.com/bcentral/prep04.armx
>
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RE: [Asterisk-Users] Searching the archives - new engine demo

[Girish Gopinath]

Hi,

Very good! I tried some of the features and they are really good, especially 
the search within selected months. I have been reading this list for the 
last 4 months. Answers for some(all) of my doubts are there in the mails 
posted within these months. Normally when googling, I try different 
combinations to find out these mails. It is ok, but time consuming. Your 
search engine is just perfect for people like me and i hope someone will 
host it.

Regards...

Girish

From: Kim Hendrikse <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Searching the archives - new engine demo
Date: Sat, 24 Jan 2004 13:21:13 +0100
Hi,

I've placed a demo search engine of the asterisk users archive here:

http://asterisk.nextrieve.com/cgi-bin/asterisk

I know there are a number of other ways to search this list that have been
suggested and one person suggested that another wasn't necessary but this
engine will do some things that the others can't. Specifically you can 
search
within selected months and from specific users. Or require a certain word 
in a
subject. You also do a fuzzy search if you are not sure of the spelling, 
which
approximates a phrase search.

I'll leave this demo up for a couple of days, if this is interesting to 
people
and someone wishes to host it I can provide the code.

I've noticed myself that it can be difficult to search the list within 
certain
time periods. Google simply won't do that. Not base upon the time the mail 
was
sent.

Make sure you check out the advance features which is where you can 
restrict
to sender, exclude words etc.

  - Kim Hendrikse
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[Asterisk-Users] IAX hard phone

[Robert Boardman]

Has any one seen or heard of the lastest developments fo the Farfon IAX 
phone? the web site

Thanks

Robb

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Re: [Asterisk-Users] 8 lines - best approach

[David Liu]

Title: RE: [Asterisk-Users] 8 lines - best approach



Vegastream is a good choice.  And it is 
tested to work with Primus' SIP platform.  Only $2212 with 10 
FXO.
 
David

  - Original Message - 
  From: 
  Kostur, 
  Andre 
  To: '[EMAIL PROTECTED]' 
  
  Sent: Saturday, January 24, 2004 12:06 
  AM
  Subject: RE: [Asterisk-Users] 8 lines - 
  best approach
  
  Hey neighbour!   I'll be posting on here what sort 
  of experience we'll have with 8 (actually 10) incoming FXO lines going to a 
  Vegastream gateway...
  > -Original Message- > 
  From: Darren Martz [mailto:[EMAIL PROTECTED]] 
  > Sent: Friday, January 23, 2004 6:10 PM > To: Chris Albertson; [EMAIL PROTECTED] 
  > Subject: Re: [Asterisk-Users] 8 lines - best 
  approach > > 
  > Thanks Chris!! > 
  > Running copper does not seem logical to me 
  either. The last > time I checked > (Sept03), the cost of 8 lines in a T1 was almost double the 
  > cost of split > 
  lines. > > I have been 
  considering NuFone, and have investigated it. We > 
  have also been > looking for a decent IP based 
  business phone, but I'll post a separate > question 
  for that :) > > I have 
  three businesses all with the same problem. A decent > T1 price would > be the best, so I 
  could centralize everything and only > outsource 
  the long > distance side. > > I'm in Vancouver, BC Canada with the 
  "Telus Inc" monopoly. > What range of prices do 
  most American telco's charge for a T1?? > 
  > The price (in C$) I was quoted was $450/m plus 
  $27/m per > voice channel with > zero features. Plus there was a $1200 setup fee. > > > - 
  Original Message - > From: "Chris Albertson" 
  <[EMAIL PROTECTED]> > To: "Darren 
  Martz" <[EMAIL PROTECTED]> > Sent: Friday, 
  January 23, 2004 8:49 AM > Subject: Fwd: 
  [Asterisk-Users] 8 lines - best approach > 
  > > > > First get rid of those 8 analog 
  lines.  then you'l have > two options: 
  > > 1) Have the local phone 
  company provide you with a T1 line > that you can 
  plug directly into the Digium card.  After all > it seems silly for ther phone company to split out the 
  lines > to 8 pairs runs 16 coppr wires only to have 
  you re-combine > them. > 
  > 2) Get 8 DID numbers from a VOIP provider like 
  NuFone or > Iconnect and have all your incomming 
  calls come in over > your Internet link.  Now 
  yu've got zero hardawre, except for > the 
  PC. > > I suppose you 
  would want local extensions...  Depending on > 
  the numbr you might want a channel bank and anlog desk phones > or go with all IP Phones > 
  > > > > --- Darren Martz 
  <[EMAIL PROTECTED]> wrote: > > From: 
  "Darren Martz" <[EMAIL PROTECTED]> > > 
  To: <[EMAIL PROTECTED]> > > 
  Subject: [Asterisk-Users] 8 lines - best approach > 
  > Date: Fri, 23 Jan 2004 07:30:42 -0800 > 
  > > > I have 8 lines coming into an existing 
  PBX system and am looking for > > a cost 
  > > effective way to replace the existing system with 
  Asterisk. We need > > some of > > the features in Asterisk, including its ability to support 
  remote > > offices > 
  > (long distance savings). > > 
  > > At first glance this appears to require a T100P 
  card and a channel > > bank, but 
  > > that seems rather expensive. My estimated price on 
  that would be > > roughly > > $2600 for 8 lines given that system - perhaps my estimate is 
  way > > off > 
  > > > Is there another way that is more cost 
  effective? > > > > 
  ___ > 
  > Asterisk-Users mailing list > > 
  [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users 
  > > To UNSUBSCRIBE or update options visit: 
  > >    http://lists.digium.com/mailman/listinfo/asterisk-users 
  > > > = > Chris Albertson 
  >   Home:   310-376-1029  
  [EMAIL PROTECTED] >   
  Cell:   310-990-7550 >   
  Office: 310-336-5189  [EMAIL PROTECTED] >   KG6OMK > > __ > Do 
  you Yahoo!? > Yahoo! SiteBuilder - Free web site 
  building tool. Try it! > http://webhosting.yahoo.com/ps/sb/ > > 
  ___ > 
  Asterisk-Users mailing list > 
  [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users 
  > To UNSUBSCRIBE or update options visit: >    http://lists.digium.com/mailman/listinfo/asterisk-users 
  > 

[Asterisk-Users] Searching the archives - new engine demo

[Kim Hendrikse]

Hi,

I've placed a demo search engine of the asterisk users archive here:

http://asterisk.nextrieve.com/cgi-bin/asterisk

I know there are a number of other ways to search this list that have been
suggested and one person suggested that another wasn't necessary but this
engine will do some things that the others can't. Specifically you can search
within selected months and from specific users. Or require a certain word in a
subject. You also do a fuzzy search if you are not sure of the spelling, which
approximates a phrase search.

I'll leave this demo up for a couple of days, if this is interesting to people
and someone wishes to host it I can provide the code.

I've noticed myself that it can be difficult to search the list within certain
time periods. Google simply won't do that. Not base upon the time the mail was
sent.

Make sure you check out the advance features which is where you can restrict
to sender, exclude words etc.

  - Kim Hendrikse
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Re: [Asterisk-Users] Asterisk + BudgeTone (behind NAT)

[info-lists]

John Todd said:
>
> Time to dump the Netgear router.  That's an unacceptable answer for a
> router vendor to say "Oh, well, for this MAJOR protocol we're going
> to simply corrupt those packets so they're unusable."  What!?
>
> JT
> __

OR get an older one from eBay.
Sounds like in Netgear broke something in the newer routers.  I have an
RT311 (around 5 years old) that has been working great (knock on wood). 
My SIP phones work fine with FWD using an outbound proxy.  Havn't applied
the SIP patch to Asterisk yet but from what I've read that will solve the
Asterisk SIP over the router problem.  Need to do that anyhow in order to
do the ENUM lookup tests.

Had thought about getting a new "Modern" Netgear router but guess I'll
just keep my trusty old one!!!

Robert
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Re: [Asterisk-Users] rc.local dont works

[Karsten Wemheuer]

Hi Miklos,

listas iPfone wrote:
> Hi ! thanks for the answer..
> 
> I use rh9...

Sorry, I am familiar with Linux From Scratch, Debian and Gentoo but not
with RH.

> 
> > I think with an interrupt problem, any startup will fail, may it be
> > manual or automatic during startup.
> 
> but.. you think that there is a problem in the interrupts at all? i don´t
> understand.

Sorry, that was a little bit irritating from me. English is not my
mother tongue. I mean, if it is an interrupt problem, there would be no
difference in the results. So, I think it is NOT an interrupt problem.

HTH & HAND
Karsten


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[Asterisk-Users] strange problem with grandstream software 1.0.4.39

[Roy Sigurd Karlsbakk]

hi all

I have a strange problem that started right after an upgrade from 
1.0.3.81: Every now and then the display flashes 484 when the phone is 
idle, on hook. Early Dial is disabled, and I don't understand anything. 
Everything works fine apart from this annoying flashing...

Anyone that knows what this might be?

regards

roy

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[Asterisk-Users] Asterisk and Mandrake

[Mike Nash]

Thanks to Dave and Matt for your help.  I now have Asterisk running on my 
Mandrake 9.1 box - time to learn how to configure it :)

Regards

Mike

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Re: [Asterisk-Users] Problem installing Asterisk with Mandrake 9.1

[Mike Nash [Tall Emu]]

Thanks Dave, I will give that a try!

Regards

Mike
- Original Message - 
From: "Dave Cotton" <[EMAIL PROTECTED]>
To: "Asterisk List" <[EMAIL PROTECTED]>
Sent: Saturday, January 24, 2004 7:11 PM
Subject: Re: [Asterisk-Users] Problem installing Asterisk with Mandrake 9.1


> On Sat, 2004-01-24 at 08:25, Mike Nash [Tall Emu] wrote:
> 
> >  make: bison: Command not found
> >  make: *** [ast_expr.c] Error 127
> 
> IIRC Mandrake 9.1 does not have Bison it has Bison++ or something, I had
> the same problems and took a real Bison rpm from an earlier release.

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[Asterisk-Users] MSN 6.1

[Soragan]

Hi,

Anybody has clue of what protocol is using in MSN 6.1? SIP?

Thanks

 

Regards,

 

Soragan

Re: [Asterisk-Users] retrans_pkt: Maximum retries exceeded on call

[Kannaiyan Natesan]

You are having Cisco 7960G behind NAT.
 
Try with  nat=yes
 
I'm not sure any other settings will solve that in 
asterisk.
I have tried but no luck.
 
Kannaiyan
 

  - Original Message - 
  From: 
  Chris Wilson 
  
  To: [EMAIL PROTECTED] 
  
  Sent: Saturday, January 24, 2004 8:26 
  AM
  Subject: [Asterisk-Users] retrans_pkt: 
  Maximum retries exceeded on call
  
  Hey,
   
  I'm getting an odd message in my logs, and 
  have'nt been able to find much information on it:
   
  Jan 24 00:22:39 WARNING[-1137431632]: 
  chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] 
  for seqno 102 (Request)
  
  I'm running asterisk with a Cisco 7960G 
  
   
  If anyone know's why i'd get this.Any help 
  would be appreciated! =] Thanks! 
   
  Chris

[Asterisk-Users] Is there any plans for Digium ISDN BRI card?

[Anton Tinchev]

Yes, i know that there are many ISDN card on the market.
But when i spend money for ISDN card, i prefer to be Digiums, to get all 
support and help Asterisk :).

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Re: [Asterisk-Users] RFC3389 support issue with DG104S

[Kannaiyan Natesan]

Why don't we make RFC3389 support complete.
Is there is any progress around on that?

Kannaiyan

- Original Message - 
From: "Brian West" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, January 24, 2004 1:49 AM
Subject: Re: [Asterisk-Users] RFC3389 support issue with DG104S


> Its silence supression.  Turn that off and it will stop doing that.
> 
> bkw
> 
> On Fri, 23 Jan 2004, Zot O'Connor wrote:
> 
> > I am getting (with older image):
> >
> > RFC3389 support incomplete.  Turn off on client if possible
> >
> >
> > How do I turn that off for the DG104s?  Or if I can't how do I tweak
> > asterisk?
> >
> > I see posts about ATA-186's having an audiomode, but the closet I came
> > to was inbanddtmf.  I tried =0 and =1, no effect.
> >
> > Thanks!
> >
> >
> > --
> > Zot O'Connor <[EMAIL PROTECTED]>
> > White Knight Hackers, Inc.
> >
> > ___
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> 
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[Asterisk-Users] retrans_pkt: Maximum retries exceeded on call

[Chris Wilson]

Hey,
 
I'm getting an odd message in my logs, and have'nt 
been able to find much information on it:
 
Jan 24 00:22:39 WARNING[-1137431632]: 
chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] 
for seqno 102 (Request)

I'm running asterisk with a Cisco 7960G 

 
If anyone know's why i'd get this.Any help 
would be appreciated! =] Thanks! 
 
Chris

Re: [Asterisk-Users] Latest cvs * compile error anyone?

[WipeOut]

Kannaiyan Natesan wrote:

If you are not users from mysql database then you can disable in the
makefile.
For this,

USE_MYSQL_FRIENDS=1

change it to

USE_MYSQL_FRIENDS=0

You won't get that error.

Alternatively you can install mysqlclient library to compile it without
errors.
Kannaiyan

 

What is MYSQL_FRIENDS?? Has some new feature been added to Asterisk to 
support MySQL?

Later..

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Re: [Asterisk-Users] Problem installing Asterisk with Mandrake 9.1

[Dave Cotton]

On Sat, 2004-01-24 at 08:25, Mike Nash [Tall Emu] wrote:

>  make: bison: Command not found
>  make: *** [ast_expr.c] Error 127

IIRC Mandrake 9.1 does not have Bison it has Bison++ or something, I had
the same problems and took a real Bison rpm from an earlier release.

-- 
Dave Cotton <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] 8 lines - best approach

[Kostur, Andre]

Title: RE: [Asterisk-Users] 8 lines - best approach





Hey neighbour!   I'll be posting on here what sort of experience we'll have with 8 (actually 10) incoming FXO lines going to a Vegastream gateway...

> -Original Message-
> From: Darren Martz [mailto:[EMAIL PROTECTED]]
> Sent: Friday, January 23, 2004 6:10 PM
> To: Chris Albertson; [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] 8 lines - best approach
> 
> 
> Thanks Chris!!
> 
> Running copper does not seem logical to me either. The last 
> time I checked
> (Sept03), the cost of 8 lines in a T1 was almost double the 
> cost of split
> lines.
> 
> I have been considering NuFone, and have investigated it. We 
> have also been
> looking for a decent IP based business phone, but I'll post a separate
> question for that :)
> 
> I have three businesses all with the same problem. A decent 
> T1 price would
> be the best, so I could centralize everything and only 
> outsource the long
> distance side.
> 
> I'm in Vancouver, BC Canada with the "Telus Inc" monopoly.
> What range of prices do most American telco's charge for a T1??
> 
> The price (in C$) I was quoted was $450/m plus $27/m per 
> voice channel with
> zero features. Plus there was a $1200 setup fee.
> 
> 
> - Original Message - 
> From: "Chris Albertson" <[EMAIL PROTECTED]>
> To: "Darren Martz" <[EMAIL PROTECTED]>
> Sent: Friday, January 23, 2004 8:49 AM
> Subject: Fwd: [Asterisk-Users] 8 lines - best approach
> 
> 
> 
> 
> First get rid of those 8 analog lines.  then you'l have
> two options:
> 
> 1) Have the local phone company provide you with a T1 line
> that you can plug directly into the Digium card.  After all
> it seems silly for ther phone company to split out the lines
> to 8 pairs runs 16 coppr wires only to have you re-combine
> them.
> 
> 2) Get 8 DID numbers from a VOIP provider like NuFone or
> Iconnect and have all your incomming calls come in over
> your Internet link.  Now yu've got zero hardawre, except for
> the PC.
> 
> I suppose you would want local extensions...  Depending on
> the numbr you might want a channel bank and anlog desk phones
> or go with all IP Phones
> 
> 
> 
> 
> --- Darren Martz <[EMAIL PROTECTED]> wrote:
> > From: "Darren Martz" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Subject: [Asterisk-Users] 8 lines - best approach
> > Date: Fri, 23 Jan 2004 07:30:42 -0800
> >
> > I have 8 lines coming into an existing PBX system and am looking for
> > a cost
> > effective way to replace the existing system with Asterisk. We need
> > some of
> > the features in Asterisk, including its ability to support remote
> > offices
> > (long distance savings).
> >
> > At first glance this appears to require a T100P card and a channel
> > bank, but
> > that seems rather expensive. My estimated price on that would be
> > roughly
> > $2600 for 8 lines given that system - perhaps my estimate is way
> > off
> >
> > Is there another way that is more cost effective?
> >
> > ___
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> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> =
> Chris Albertson
>   Home:   310-376-1029  [EMAIL PROTECTED]
>   Cell:   310-990-7550
>   Office: 310-336-5189  [EMAIL PROTECTED]
>   KG6OMK
> 
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Re: [Asterisk-Users] Problem installing Asterisk with Mandrake 9.1

[Mike Nash [Tall Emu]]

Thanks Matt,

 Thanks for the very quick response.  The reason I had thought it was an
error is because of the

 "> > cc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g-c -o
"
 lines, and my subsequent inability to compile Asterisk.  Thanks for the
heads up on that, I know what to check now, but I am not in front of the box
:(

 Anyhow, looks like I am seeking my error in the wrong place - when I tried
to compile Asterisk it "definately" didn't work - I get missing
files/directories, followed ultimately by

 make: bison: Command not found
 make: *** [ast_expr.c] Error 127

 I'd appreciate any insight you may be able to give.

Regards


 Mike


 [EMAIL PROTECTED] asterisk]# make clean;
  -- seemed all ok
[EMAIL PROTECTED] asterisk]# make install
[EMAIL PROTECTED] asterisk]# make install
./mkdep -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarat
ions -g  -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE  -O6 -march=i686
-DZ
APTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-01/23/04-21:49:01\"
-DINSTALL_PREFI
X=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\"
-DASTVARL
IBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\"
-DASTSPOOLDIR=\"/var/spo
ol/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\"
-DASTCONFPATH=\"/etc/asterisk/as
terisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\"
-DASTAGIDIR=\"/var/lib/a
sterisk/agi-bin\" -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP  `ls *.c`
cli.c:31:19: build.h: No such file or directory
dlfcn.c:40:25: mach-o/dyld.h: No such file or directory
dlfcn.c:41:26: mach-o/nlist.h: No such file or directory
dlfcn.c:42:28: mach-o/getsect.h: No such file or directory
for x in res channels pbx apps codecs formats agi cdr astman stdtime; do
make
-C
 $x depend || exit 1 ; done
make[1]: Entering directory `/usr/src/asterisk/res'
../mkdep -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declara
tions -g  -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE  -O6 -march=i686
-D
ZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-01/23/04-21:49:01\"
-DINSTALL_PREF
IX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\"
-DASTVAR
LIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\"
-DASTSPOOLDIR=\"/var/sp
ool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\"
-DASTCONFPATH=\"/etc/asterisk/a
sterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\"
-DASTAGIDIR=\"/var/lib/
asterisk/agi-bin\" -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP
-DZAPATA_MOH
-DOPENSSL_NO_KRB5 -fPIC `ls *.c`
make[1]: Leaving directory `/usr/src/asterisk/res'
make[1]: Entering directory `/usr/src/asterisk/channels'
../mkdep -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declara
tions -g  -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE  -O6 -march=i686
-D
ZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-01/23/04-21:49:01\"
-DINSTALL_PREF
IX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\"
-DASTVAR
LIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\"
-DASTSPOOLDIR=\"/var/sp
ool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\"
-DASTCONFPATH=\"/etc/asterisk/a
sterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\"
-DASTAGIDIR=\"/var/lib/
asterisk/agi-bin\" -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP
-Wno-missing-pr
ototypes -Wno-missing-declarations   -DZAPATA_PRI   -DIAX_TRUNKING -DCRYPTO
-fPI
C  `ls *.c`
chan_alsa.c:29:28: alsa/asoundlib.h: No such file or directory
chan_alsa.c:30:18: busy.h: No such file or directory
chan_alsa.c:31:22: ringtone.h: No such file or directory
chan_nbs.c:33:17: nbs.h: No such file or directory
chan_oss.c:43:18: busy.h: No such file or directory
chan_oss.c:44:22: ringtone.h: No such file or directory
chan_vpb.c:34:20: vpbapi.h: No such file or directory
make[1]: Leaving directory `/usr/src/asterisk/channels'
make[1]: Entering directory `/usr/src/asterisk/pbx'
../mkdep -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declara
tions -g  -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE  -O6 -march=i686
-D
ZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-01/23/04-21:49:01\"
-DINSTALL_PREF
IX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\"
-DASTVAR
LIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run\"
-DASTSPOOLDIR=\"/var/sp
ool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\"
-DASTCONFPATH=\"/etc/asterisk/a
sterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\"
-DASTAGIDIR=\"/var/lib/
asterisk/agi-bin\" -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP  -fPIC `ls
*.c`
pbx_gtkconsole.c:38:21: gtk/gtk.h: No such file or directory
pbx_gtkconsole.c:39:18: glib.h: No such file or directory
make[1]: Leaving directory `/usr/src/asterisk/pbx'
make[1]: Entering directory `/usr/src/asterisk/apps'
../mkdep -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declara
tions -g  -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE  -O6 -march=i686
-D
ZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-01/23/04-21:49:01\"
-DINSTALL_PREF
IX=\"\" -DASTETCDIR=\"/etc/asterisk\" -D

Re: Subject: Re: [Asterisk-Users] Grandstream 100 sidetone

[Chris Albertson]

--- dkwok <[EMAIL PROTECTED]> wrote:
> Chris Albertson wrote:
> 
> |What firmware version do you have?
> 
> program version 1.0.4.39

I've got the same firmware version.  So it appears that sidetone
volume is not dependent on the firmware version.  


> 
> -- 
> David Kwok
> 
> Iaxtel/FWD # 17001813482 ext 1002
> 

> ATTACHMENT part 2 application/x-pkcs7-signature name=smime.p7s



=
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  Home:   310-376-1029  [EMAIL PROTECTED]
  Cell:   310-990-7550
  Office: 310-336-5189  [EMAIL PROTECTED]
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Re: Subject: Re: [Asterisk-Users] Grandstream 100 sidetone

[Chris Albertson]

--- dkwok <[EMAIL PROTECTED]> wrote:
> Chris Albertson wrote:
> 
> |What firmware version do you have?
> 
> program version 1.0.4.39

I've got the same firmware version.  So it appears that sidetone
volume is not dependent on the firmware version.  


> 
> -- 
> David Kwok
> 
> Iaxtel/FWD # 17001813482 ext 1002
> 

> ATTACHMENT part 2 application/x-pkcs7-signature name=smime.p7s



=
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  Office: 310-336-5189  [EMAIL PROTECTED]
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