Re: [Asterisk-Users] UK Caller ID and X100p

[Linus Surguy]

> Hi
> I really need caller id to work in the UK, I understand that the X100p
> uses a US chipset,two questions
> 1) is that a product that converts UK to US caller id in line

Not really the answer you were looking for, but if you get a line from a
non-BT supplier (e.g. NTL or Telewest) you are quite likely to find that CLI
works.

Linus

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Re: [Asterisk-Users] delay problem in h323

[andrea]

thanks, I'll try. Question: asterisk always manages RTP flow also with 
chan_h323?

Andrea

Steven Thomas wrote:





Hi,

I use Asterisk as a SIP <-> H323 translator without any issues after
switching to chan_h323.
My environment is:

SIP (7960) -> Asterisk -> GnuGK (h323) -> Cisco 2600 H323 Gateway to PSTN.

This works well without the CPU load seen with oh323.  The call control
also seems far better using chan_h323.  I have no delay either.
I use a smaller box: PII 200, 64Mb RAM.  RedHat 9.

Only 4 handsets - 2 SIP IP 7960's, 2 Analog H323 via the Cisco router FXS
ports.
I also have configured Asterisk on another site to act as a H323 gateway
for PSTN calls into a Cisco Call Manager via gnuGK - H323 also.
I would suggest trying chan_h323 as an alternative.



Regards,

Steven Thomas

Technical Project Manager
Network & Connectivity  Services, IBM Australia
Ph: 0404 099 262
NH011, IBM Centre, St Leonards, 2065
Internet:  [EMAIL PROTECTED]
Visit us at http://www.ibm.com/services/au/its



 
  andy <[EMAIL PROTECTED]>   
  Sent by:  To:   "" <[EMAIL PROTECTED]>   
  [EMAIL PROTECTED]cc:  
  .digium.com   Subject:  Re: [Asterisk-Users] delay problem in h323 
 
 
  10-09-03 08:24 AM  
  Please respond to  
  asterisk-users 
 



yes, I agree with you.
I verify with a sniffer and asterisk manages RTP flows. The problem is
asterisk
decode and then code again RTP flows. This function requires 5-7% CPU On my
test-box (Linux rh 7.3 on P3 600 GHz). This solution  don't scale without
dedicated
HW, I think!
Another problem is codec supported: ok for G.711, G.729. I don't know for
GSM
BUT: what about video codec? what about proprietary codec or ciphered
codec?
Do you have any suggestion on how I can manage this with asterisk? I'm very

interested into asterisk as sip-to-h323 translator.
Thanks
Andrea

Quoting Steven Thomas <[EMAIL PROTECTED]>:






The only way I was able to solve my delay issue with Chan_oh323 was to
switch to Chan_h323.
Chan_oh323 caused a similar 3 -4 sec delay on one way of the
conversation.

Checking the CPU stats on asterisk during the call - confirms that the
RTP

stream was somehow routing through asterisk - not sure why!



Regards,

Steven Thomas







 andrea <[EMAIL PROTECTED]>


 Sent by:  To:
[EMAIL PROTECTED]


 [EMAIL PROTECTED]cc:


 .digium.com   Subject:  Re:
[Asterisk-Users] delay problem in h323




 10-09-03 12:45 AM


 Please respond to


 asterisk-users






Hi all,

is it possible to disable RTP routing through asterisk? RTP routing is a
very nice feature but, I think it’s important also to disable it in some
cases (e. g. in a LAN).
Do you have any suggestion?
Andrea

Rattana BIV wrote:


Hi,

I have a delay between two H323.

Netmeeting1 - ||
| gnuGK | --- [asterisk-oh323]
| Asterisk |
Netmeeting2 --||
Netmmeting 1 call Netmeeting 2. When Netmmeting 1 speak Netmeeting 2
receive the voice without delay. But in the other way I have 3 secondes
delay.
In oh323.conf  I set jittermin and jittermax to 20, the ipTos=lowdelay.
I try to find where I can delete the delay.
Does anyone have a tip ?
Best Regards
Rattana


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[Asterisk-Users] Re: SIP LD carrier

[Louis-David Mitterrand]

On Tue, Sep 09, 2003 at 07:57:20PM -0400, Jeremy McNamara wrote:
> Travis Johnson wrote:
> 
> >I've called NuFone and was not impressed by their voicemail answering 
> >system (choppy) and was unable to even leave a message before the phone 
> >call was disconnected (in the middle of the
> >recording).
> > 
> So your going to judge our system by making one phone call into my home 
> asterisk system that runs on a fully saturated ADSL connection.

Wait... of course people are going to judge you by that! If putting your
company's answering machine on your (saturated) dsl connection is an
indication of your other technical choices it reflects poorly on your
company. Also your website is almost totally empty of details on your
services and pricing but heavy on doubtful eye candy.

If people on this mailing list had not time and againg recommended the
excellence of your service I wonder how you would find your customers. 

I for one sent an enquiry at [EMAIL PROTECTED] 20 hours ago and still
haven't received an answer. How hard is it to install a generic
e-mail answerer saying "here is our price list, please be patient I will
contact you"?

Now I can understand that when developping a technically ambitious
platform such as yours, communication and marketing is not your
priority, but there is no need to bash your prospective customers with
insulting remarks such as this one below:

> Typical,
^^^

-- 
 Marijuana is nature's way of saying, "Hi!".
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Re: [Asterisk-Users] Dial + disconnect

[Eric Wieling]

Yes, on ISDN PRI.  On analog you can try the busytetect and progress
detect but that always disconnects my calls at random times.

On Wed, 2003-09-10 at 00:37, Chee Foong wrote:
> Yes you are right, Sorry my mistake.
> 
> So, is there a way to detect busy, answer, or no answer call?
> 
> Foong
> 
> - Original Message - 
> From: "Richard Lyman" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Wednesday, September 10, 2003 12:50 PM
> Subject: Re: [Asterisk-Users] Dial + disconnect
> 
> 
> > based on what dial string you have a zap device '0122740900' (looks more 
> > like an exten/phone# to me) maybe you meant Zap/1/0122740900  (notice 
> > the /1/, you could also use groups /g1/ (if setup in zapata.conf))
> > 
> > Chee Foong wrote:
> > 
> > > Hello,
> > >  
> > > When I have the following extension:
> > >  
> > > exten => 900,1,dial(Zap/0122740900)
> > >  
> > > can I know whether 'dial' actually gets through or the called party is 
> > > busy at the moment. I want to perform different action based on 
> > > whether the 'dail' success or not.
> > >  
> > >  
> > > Foong 
> > 
> > 
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> ___
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-- 
BTEL Consulting
850-484-4535 x2111 (Office)
504-595-3916 x2111 (Experimental)
877-552-0838 (Backup Phone)

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Re: [Asterisk-Users] Dial + disconnect

[Chee Foong]

Yes you are right, Sorry my mistake.

So, is there a way to detect busy, answer, or no answer call?

Foong

- Original Message - 
From: "Richard Lyman" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, September 10, 2003 12:50 PM
Subject: Re: [Asterisk-Users] Dial + disconnect


> based on what dial string you have a zap device '0122740900' (looks more 
> like an exten/phone# to me) maybe you meant Zap/1/0122740900  (notice 
> the /1/, you could also use groups /g1/ (if setup in zapata.conf))
> 
> Chee Foong wrote:
> 
> > Hello,
> >  
> > When I have the following extension:
> >  
> > exten => 900,1,dial(Zap/0122740900)
> >  
> > can I know whether 'dial' actually gets through or the called party is 
> > busy at the moment. I want to perform different action based on 
> > whether the 'dail' success or not.
> >  
> >  
> > Foong 
> 
> 
> ___
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> 
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Re: [Asterisk-Users] Nortel i2004 and asterisk ??

[Dave Packham]

several SIP webpages list the 2004i as a SIP hardphone. I have 50+ of them that I 
would love to use with * but cant one Nortel rep said they are writing a flash for 
it and one said not.  we can only hope

Dave



>>> [EMAIL PROTECTED] 9/9/2003 10:09:23 PM >>>
Where did you hear that they were SIP Phones??  Its not true.

What is true is that there is a software called the "Nortel Networks SIP 
Client".  This is a windows PC SIP Software that has the ability to control 
the i2004, so in turn the i2004 behaves as a SIP phone (provided the Windows 
PC is running).

You will have to contact Nortel directly to get the software.  Otherwise the 
i2004 only works with the Nortel BCM and Succession Equipment.


On Tuesday 09 September 2003 21:06, John Brown wrote:
> This are "suppose" to be SIP based phones.  I just
> got one to play with, no docs, no nothing.
>
> before I start spending time trying to sort it
> out I thought I'd ask to see if anyone had made
> one work with AST, and if they have any pearls
> of wisdom.
>
> mucho thanks
>
>
> ___
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Re: [Asterisk-Users] Transfer of queue call

[Richard Lyman]

*CLI> show application Queue

 -= Info about application 'Queue' =-

[Synopsis]:
 Queue a call for a call queue
[Description]:
 Queue(queuename[|options[|URL][|announceoverride]]):
Queues an incoming call in a particular call queue as defined in 
queues.conf.
 This application returns -1 if the originating channel hangs up, or if the
call is bridged and  either of the parties in the bridge terminate the call.
Returns 0 if the queue is full, nonexistant, or has no members.
The option string may contain zero or more of the following characters:
 't' -- allow the called user transfer the calling user
 'T' -- to allow the calling user to transfer the call.
 'd' -- data-quality (modem) call (minimum delay).
 'H' -- allow caller to hang up by hitting *.
 In addition to transferring the call, a call may be parked and then picked
up by another user.
 The optionnal URL will be sent to the called party if the channel supports
it.

Hielke Christian Braun wrote:

Hello,

hope somebody can help. I have setup a queue which maps to some
Budgetone SIP phones. When a call is answered, the # key to transfer
a call does not work. Everything else regarding the queue works fine.
Is there a way to activate it? Maybe something like the t option in 
the Dial application.

Thanks in advance,
Christian. 

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Re: [Asterisk-Users] Dial + disconnect

[Richard Lyman]

based on what dial string you have a zap device '0122740900' (looks more 
like an exten/phone# to me) maybe you meant Zap/1/0122740900  (notice 
the /1/, you could also use groups /g1/ (if setup in zapata.conf))

Chee Foong wrote:

Hello,
 
When I have the following extension:
 
exten => 900,1,dial(Zap/0122740900)
 
can I know whether 'dial' actually gets through or the called party is 
busy at the moment. I want to perform different action based on 
whether the 'dail' success or not.
 
 
Foong 


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Re: [Asterisk-Users] Nortel i2004 and asterisk ??

[Andres]

Where did you hear that they were SIP Phones??  Its not true.

What is true is that there is a software called the "Nortel Networks SIP 
Client".  This is a windows PC SIP Software that has the ability to control 
the i2004, so in turn the i2004 behaves as a SIP phone (provided the Windows 
PC is running).

You will have to contact Nortel directly to get the software.  Otherwise the 
i2004 only works with the Nortel BCM and Succession Equipment.


On Tuesday 09 September 2003 21:06, John Brown wrote:
> This are "suppose" to be SIP based phones.  I just
> got one to play with, no docs, no nothing.
>
> before I start spending time trying to sort it
> out I thought I'd ask to see if anyone had made
> one work with AST, and if they have any pearls
> of wisdom.
>
> mucho thanks
>
>
> ___
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[Asterisk-Users] Transfer of queue call

[Hielke Christian Braun]

Hello,

hope somebody can help. I have setup a queue which maps to some
Budgetone SIP phones. When a call is answered, the # key to transfer
a call does not work. Everything else regarding the queue works fine.

Is there a way to activate it? Maybe something like the t option in 
the Dial application.


Thanks in advance,
 Christian. 

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[Asterisk-Users] Dial + disconnect

[Chee Foong]

Hello,
 
When I have the following extension:
 
exten => 900,1,dial(Zap/0122740900)
 
can I know whether 'dial' actually gets through or 
the called party is busy at the moment. I want to perform different action based 
on whether the 'dail' success or not.
 
 
Foong 

Re: [Asterisk-Users] delay problem in h323

[Steven Thomas]

Hi,

I use Asterisk as a SIP <-> H323 translator without any issues after
switching to chan_h323.

My environment is:

SIP (7960) -> Asterisk -> GnuGK (h323) -> Cisco 2600 H323 Gateway to PSTN.

This works well without the CPU load seen with oh323.  The call control
also seems far better using chan_h323.  I have no delay either.

I use a smaller box: PII 200, 64Mb RAM.  RedHat 9.

Only 4 handsets - 2 SIP IP 7960's, 2 Analog H323 via the Cisco router FXS
ports.

I also have configured Asterisk on another site to act as a H323 gateway
for PSTN calls into a Cisco Call Manager via gnuGK - H323 also.

I would suggest trying chan_h323 as an alternative.



Regards,

Steven Thomas


Technical Project Manager
Network & Connectivity  Services, IBM Australia

Ph: 0404 099 262
NH011, IBM Centre, St Leonards, 2065
Internet:  [EMAIL PROTECTED]

Visit us at http://www.ibm.com/services/au/its



   
  
  andy <[EMAIL PROTECTED]> 
  
  Sent by:  To:   "" <[EMAIL 
PROTECTED]>   
  [EMAIL PROTECTED]cc: 
 
  .digium.com   Subject:  Re: [Asterisk-Users] 
delay problem in h323 
   
  
   
  
  10-09-03 08:24 AM
  
  Please respond to
  
  asterisk-users   
  
   
  



yes, I agree with you.
I verify with a sniffer and asterisk manages RTP flows. The problem is
asterisk
decode and then code again RTP flows. This function requires 5-7% CPU On my

test-box (Linux rh 7.3 on P3 600 GHz). This solution  don't scale without
dedicated
HW, I think!

Another problem is codec supported: ok for G.711, G.729. I don't know for
GSM
BUT: what about video codec? what about proprietary codec or ciphered
codec?

Do you have any suggestion on how I can manage this with asterisk? I'm very

interested into asterisk as sip-to-h323 translator.
Thanks

Andrea


Quoting Steven Thomas <[EMAIL PROTECTED]>:

>
>
>
>
>
> The only way I was able to solve my delay issue with Chan_oh323 was to
> switch to Chan_h323.
>
> Chan_oh323 caused a similar 3 -4 sec delay on one way of the
conversation.
> Checking the CPU stats on asterisk during the call - confirms that the
RTP
> stream was somehow routing through asterisk - not sure why!
>
>
>
> Regards,
>
> Steven Thomas
>
>
>
>
>

>
>   andrea <[EMAIL PROTECTED]>

>
>   Sent by:  To:
> [EMAIL PROTECTED]

>   [EMAIL PROTECTED]cc:

>
>   .digium.com   Subject:  Re:
> [Asterisk-Users] delay problem in h323
>

>
>

>
>   10-09-03 12:45 AM

>
>   Please respond to

>
>   asterisk-users

>
>

>
>
>
>
> Hi all,
>
> is it possible to disable RTP routing through asterisk? RTP routing is a
> very nice feature but, I think it’s important also to disable it in some
> cases (e. g. in a LAN).
> Do you have any suggestion?
>
> Andrea
>
> Rattana BIV wrote:
>
> > Hi,
> >
> > I have a delay between two H323.
> >
> > Netmeeting1 - ||
> >  | gnuGK | --- [asterisk-oh323]
> > | Asterisk |
> > Netmeeting2 --||
> >
> > Netmmeting 1 call Netmeeting 2. When Netmmeting 1 speak Netmeeting 2
> > receive the voice without delay. But in the other way I have 3 secondes
> > delay.
> > In oh323.conf  I set jittermin and jittermax to 20, the ipTos=lowdelay.
> > I try to find where I can delete the delay.
> > Does anyone have a tip ?
> >
> >
> > Best Regards
> > Rattana
> >
>
>
>
> ___
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> [EMAIL PROTECTED]
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> Ë^®+$RÇ«²f¢–)à–+-Ë^®+$RÇ«²X¬¶Çb‚+Šr‰¡¶ÚþX¬¶Çb‚+Šr‰¿™š¥™©ÿ–+-Šwèý«-
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Re: [Asterisk-Users] BudgeTone-100 Early Dial

[Hielke Christian Braun]

> On Tue, 2003-09-09 at 16:03, Tilghman Lesher wrote:
> > Why not just use DISA:
> > 
> > exten => 9,1,DISA(no-password|outgoing)
> 
> Because I didn't know about it. :)  I'll try it out.
> 

Thanks, the workaround with the dialtone works great. Just had some
trouble getting a wav file of a dialtone. If anybody needs it, i put
it on my site: http://www.unco.de/asterisk/dialtone.wav

The problem i had with the DISA option is that transfers with the #
did not work anymore.  


Christian.
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Re: [Asterisk-Users] Endpoint-to-Endpoint RTP Packets

[Mike Ciholas]

On Tue, 9 Sep 2003, Eric Wieling wrote:

> It would have to do some kind of trascoding,

Forgive my ignorance, but why?  PSTN is delivering 8 bit 8 KHz 
ulaw samples.  G711 is delivering 8 bit 8 KHz ulaw samples over 
SIP.  Aren't the two data streams identical down to the bit 
level?

-- 
Mike Ciholas(812) 476-2721 voice
CIHOLAS Enterprises (812) 476-2881 fax
2626 Kotter Ave, Unit D [EMAIL PROTECTED]
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[Asterisk-Users] Nortel i2004 and asterisk ??

[John Brown]

This are "suppose" to be SIP based phones.  I just
got one to play with, no docs, no nothing.

before I start spending time trying to sort it
out I thought I'd ask to see if anyone had made 
one work with AST, and if they have any pearls 
of wisdom.

mucho thanks


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Re: [Asterisk-Users] help on MOH config, pretty close?

[CW_ASN]

> > The source code for your kernel is not the same as the kernel actually
> > running on the machine.
I don't see that. Yes, sure.


- Original Message -
From: "CW_ASN" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, September 09, 2003 10:00 PM
Subject: Re: [Asterisk-Users] help on MOH config, pretty close?


> Sometimes it happened that error to me. Reinitiating Linux and proving
again
> (modprobe/insmod), the error disappeared.
>
>
> - Original Message -
> From: "Eric Wieling" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Tuesday, September 09, 2003 9:48 PM
> Subject: Re: [Asterisk-Users] help on MOH config, pretty close?
>
>
> > The source code for your kernel is not the same as the kernel actually
> > running on the machine.
> >
> > On Tue, 2003-09-09 at 20:38, Rich Adamson wrote:
> > > After following your suggestions, I get
> > >
> > > [EMAIL PROTECTED] zaptel]# /sbin/modprobe ztdummy
> > > /lib/modules/2.4.20-18.9/misc/zaptel.o: kernel-module version mismatch
> > > /lib/modules/2.4.20-18.9/misc/zaptel.o was compiled for kernel
> version 2
> > > .4.20-6
> > > while this kernel is version 2.4.20-18.9.
> > > /lib/modules/2.4.20-18.9/misc/zaptel.o: insmod
> /lib/modules/2.4.20-18.9/misc/zap
> > > tel.o failed
> > > /lib/modules/2.4.20-18.9/misc/zaptel.o: insmod ztdummy failed
> > > [EMAIL PROTECTED] zaptel]#
> > >
> > > I should have two X100P Digium cards arriving tomorrow. Rather then
mess
> with this,
> > > should I just wait until those cards are installed?
> > >
> > >
> > > 
> > > > If I'm not mistaken, ztdummy makes work moh and meetme...
> > > > Do lsmod from sh. If you don't see ztdummy, download zaptel, and
> modify
> > > > Makefile. Line 89 must be:
> > > >
> > > > MODULES=zaptel.o tor2.o torisa.o wcusb.o wcfxo.o wcfxs.o \
> > > > ztdynamic.o ztd-eth.o wct1xxp.o wct4xxp.o ztdummy.o
> > > >
> > > > (Remove # preceding ztdummy.o)
> > > >
> > > > Then, do a make and make install. When finish, do:
> > > >
> > > > modprobe ztdummy
> > > >
> > > > Meetme and MOH should work now.
> > > >
> > > >
> > > > - Original Message -
> > > > From: "Rich Adamson" <[EMAIL PROTECTED]>
> > > > To: <[EMAIL PROTECTED]>
> > > > Sent: Tuesday, September 09, 2003 9:59 PM
> > > > Subject: RE: [Asterisk-Users] help on MOH config, pretty close?
> > > >
> > > >
> > > > > The show modules command line suggests that is Not loaded. Should
if
> be,
> > > > and
> > > > > if so, where do I do that?
> > > > >
> > > > > Rich
> > > > >
> > > > > 
> > > > > > Do you have loaded ztdummy as module?
> > > > > >
> > > > > >
> > > > > > -Mensaje original-
> > > > > > Trying to test the music on hold function and can't seem to get
it
> to
> > > > work.
> > > > > > If anyone has it running, could you give me a clue? (I have
> googled and
> > > > > > found lots of questions, but no real suggestions.)
> > > > > >
> > > > > > I downloaded and installed the mpg123 package. From the RH9
> console I
> > > > can
> > > > > > start the executable and hear the music via the speakers. The
> executable
> > > > > > is located in /usr/bin. (That works!)
> > > > > >
> > > > > > I set the following line in musiconhold.conf file:
> > > > > >   default => mp3:/var/lib/asterisk/mohmp3/
> > > > > > and when I do a ps ax |grep mpg123 I get:
> > > > > >   15798 pts/0S  0:00 /usr/bin/mpg123 -q -s --mono -r
> 8000 -b
> > > > 2048
> > > > > > For-You.mp3
> > > > > > If I stop asterisk, the above line disappears. (That works!)
> > > > > >
> > > > > > For lab work, the sip.conf has:
> > > > > >  allow=ulaw
> > > > > > only.
> > > > > >
> > > > > > In the zapata.conf file I have:
> > > > > >   musiconhold=default
> > > > > >
> > > > > > On asterisk startup, the console window shows:
> > > > > >   == Parsing '/etc/asterisk/musiconhold.conf': Found
> > > > > >   WARNING[1074504864]: File res_musiconhold.c, Line 506
> (moh_register):
> > > > > > Unable to
> > > > > >   open pseudo channel for timing...  Sound may be choppy.
> > > > > >   == Registered application 'MusicOnHold'
> > > > > >   == Registered application 'WaitMusicOnHold'
> > > > > >   == Registered application 'SetMusicOnHold'
> > > > > > which looks like its okay.
> > > > > >
> > > > > > The the asterisk command line, I see:
> > > > > > -- Called 3002
> > > > > > -- SIP/3002-3ed3 is ringing
> > > > > > -- SIP/3002-3ed3 is ringing
> > > > > > -- SIP/3002-3ed3 answered SIP/3000-0e7b
> > > > > > -- Attempting native bridge of SIP/3000-0e7b and
SIP/3002-3ed3
> > > > > > -- Started music on hold, class 'default', on SIP/3002-3ed3
> > > > > > but nothing is heard on either C7960's or Snom 200.
> > > > > >
> > > > > > I have not modified anything in sip.conf or extensions.conf; am
I
> > > > suppose
> > > > > > to?
> > > > > >
> > > > > > What am I missing
> > > > > >
> > > > > > Rich
> > > > > >
> > > > > >
> > > > > >
> > > > > > _

Re: [Asterisk-Users] help on MOH config, pretty close?

[CW_ASN]

Sometimes it happened that error to me. Reinitiating Linux and proving again
(modprobe/insmod), the error disappeared.


- Original Message -
From: "Eric Wieling" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, September 09, 2003 9:48 PM
Subject: Re: [Asterisk-Users] help on MOH config, pretty close?


> The source code for your kernel is not the same as the kernel actually
> running on the machine.
>
> On Tue, 2003-09-09 at 20:38, Rich Adamson wrote:
> > After following your suggestions, I get
> >
> > [EMAIL PROTECTED] zaptel]# /sbin/modprobe ztdummy
> > /lib/modules/2.4.20-18.9/misc/zaptel.o: kernel-module version mismatch
> > /lib/modules/2.4.20-18.9/misc/zaptel.o was compiled for kernel
version 2
> > .4.20-6
> > while this kernel is version 2.4.20-18.9.
> > /lib/modules/2.4.20-18.9/misc/zaptel.o: insmod
/lib/modules/2.4.20-18.9/misc/zap
> > tel.o failed
> > /lib/modules/2.4.20-18.9/misc/zaptel.o: insmod ztdummy failed
> > [EMAIL PROTECTED] zaptel]#
> >
> > I should have two X100P Digium cards arriving tomorrow. Rather then mess
with this,
> > should I just wait until those cards are installed?
> >
> >
> > 
> > > If I'm not mistaken, ztdummy makes work moh and meetme...
> > > Do lsmod from sh. If you don't see ztdummy, download zaptel, and
modify
> > > Makefile. Line 89 must be:
> > >
> > > MODULES=zaptel.o tor2.o torisa.o wcusb.o wcfxo.o wcfxs.o \
> > > ztdynamic.o ztd-eth.o wct1xxp.o wct4xxp.o ztdummy.o
> > >
> > > (Remove # preceding ztdummy.o)
> > >
> > > Then, do a make and make install. When finish, do:
> > >
> > > modprobe ztdummy
> > >
> > > Meetme and MOH should work now.
> > >
> > >
> > > - Original Message -
> > > From: "Rich Adamson" <[EMAIL PROTECTED]>
> > > To: <[EMAIL PROTECTED]>
> > > Sent: Tuesday, September 09, 2003 9:59 PM
> > > Subject: RE: [Asterisk-Users] help on MOH config, pretty close?
> > >
> > >
> > > > The show modules command line suggests that is Not loaded. Should if
be,
> > > and
> > > > if so, where do I do that?
> > > >
> > > > Rich
> > > >
> > > > 
> > > > > Do you have loaded ztdummy as module?
> > > > >
> > > > >
> > > > > -Mensaje original-
> > > > > Trying to test the music on hold function and can't seem to get it
to
> > > work.
> > > > > If anyone has it running, could you give me a clue? (I have
googled and
> > > > > found lots of questions, but no real suggestions.)
> > > > >
> > > > > I downloaded and installed the mpg123 package. From the RH9
console I
> > > can
> > > > > start the executable and hear the music via the speakers. The
executable
> > > > > is located in /usr/bin. (That works!)
> > > > >
> > > > > I set the following line in musiconhold.conf file:
> > > > >   default => mp3:/var/lib/asterisk/mohmp3/
> > > > > and when I do a ps ax |grep mpg123 I get:
> > > > >   15798 pts/0S  0:00 /usr/bin/mpg123 -q -s --mono -r
8000 -b
> > > 2048
> > > > > For-You.mp3
> > > > > If I stop asterisk, the above line disappears. (That works!)
> > > > >
> > > > > For lab work, the sip.conf has:
> > > > >  allow=ulaw
> > > > > only.
> > > > >
> > > > > In the zapata.conf file I have:
> > > > >   musiconhold=default
> > > > >
> > > > > On asterisk startup, the console window shows:
> > > > >   == Parsing '/etc/asterisk/musiconhold.conf': Found
> > > > >   WARNING[1074504864]: File res_musiconhold.c, Line 506
(moh_register):
> > > > > Unable to
> > > > >   open pseudo channel for timing...  Sound may be choppy.
> > > > >   == Registered application 'MusicOnHold'
> > > > >   == Registered application 'WaitMusicOnHold'
> > > > >   == Registered application 'SetMusicOnHold'
> > > > > which looks like its okay.
> > > > >
> > > > > The the asterisk command line, I see:
> > > > > -- Called 3002
> > > > > -- SIP/3002-3ed3 is ringing
> > > > > -- SIP/3002-3ed3 is ringing
> > > > > -- SIP/3002-3ed3 answered SIP/3000-0e7b
> > > > > -- Attempting native bridge of SIP/3000-0e7b and SIP/3002-3ed3
> > > > > -- Started music on hold, class 'default', on SIP/3002-3ed3
> > > > > but nothing is heard on either C7960's or Snom 200.
> > > > >
> > > > > I have not modified anything in sip.conf or extensions.conf; am I
> > > suppose
> > > > > to?
> > > > >
> > > > > What am I missing
> > > > >
> > > > > Rich
> > > > >
> > > > >
> > > > >
> > > > > ___
> > > > > Asterisk-Users mailing list
> > > > > [EMAIL PROTECTED]
> > > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > >
> > > > > ___
> > > > > Asterisk-Users mailing list
> > > > > [EMAIL PROTECTED]
> > > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > > > ---End of Original Message-
> > > >
> > > >
> > > > ___
> > > > Asterisk-Users mailing list
> > > > [EMAIL PROTECTED]
> > > > http://lists.digiu

Re: [Asterisk-Users] freebsd and asterisk ?? anyone yet

[David Raistrick]

> another.  I mean, by the time you strip down the linux install to make it
> rock-stable, it's basically a kernel, compiler, and a handful of
> dependances. Does it really matter what OS it runs on at this point, as
> long as it's a robust kernel?

At least in my case, I /can/ strip down a FreeBSD (or a picobsd release)
install to its required components in a matter of minutes.  I keep up with
the latest security issues and major bugs.  It's a must for my daily job.
I track -stable and I'm watching -current when I can.

I feel comfortable with keeping FreeBSD secure.   To do this with Linux,
I have to start over.  I have an entirely different set of problems to
keep track of, an unfamiliar..eh..release process, etc.


If I worked in a Solaris or IRIX shop, I'd similarly prefer to run the
system in an environment closer to the "usual" working environment.

Just my thoughts. :)

...david


---
david raistrick
[EMAIL PROTECTED]   http://www.expita.com/nomime.html

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Re: [Asterisk-Users] help on MOH config, pretty close?

[Eric Wieling]

The source code for your kernel is not the same as the kernel actually
running on the machine.

On Tue, 2003-09-09 at 20:38, Rich Adamson wrote:
> After following your suggestions, I get
> 
> [EMAIL PROTECTED] zaptel]# /sbin/modprobe ztdummy
> /lib/modules/2.4.20-18.9/misc/zaptel.o: kernel-module version mismatch
> /lib/modules/2.4.20-18.9/misc/zaptel.o was compiled for kernel version 2
> .4.20-6
> while this kernel is version 2.4.20-18.9.
> /lib/modules/2.4.20-18.9/misc/zaptel.o: insmod /lib/modules/2.4.20-18.9/misc/zap
> tel.o failed
> /lib/modules/2.4.20-18.9/misc/zaptel.o: insmod ztdummy failed
> [EMAIL PROTECTED] zaptel]# 
> 
> I should have two X100P Digium cards arriving tomorrow. Rather then mess with this,
> should I just wait until those cards are installed?
> 
> 
> 
> > If I'm not mistaken, ztdummy makes work moh and meetme...
> > Do lsmod from sh. If you don't see ztdummy, download zaptel, and modify
> > Makefile. Line 89 must be:
> > 
> > MODULES=zaptel.o tor2.o torisa.o wcusb.o wcfxo.o wcfxs.o \
> > ztdynamic.o ztd-eth.o wct1xxp.o wct4xxp.o ztdummy.o
> > 
> > (Remove # preceding ztdummy.o)
> > 
> > Then, do a make and make install. When finish, do:
> > 
> > modprobe ztdummy
> > 
> > Meetme and MOH should work now.
> > 
> > 
> > - Original Message -
> > From: "Rich Adamson" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Tuesday, September 09, 2003 9:59 PM
> > Subject: RE: [Asterisk-Users] help on MOH config, pretty close?
> > 
> > 
> > > The show modules command line suggests that is Not loaded. Should if be,
> > and
> > > if so, where do I do that?
> > >
> > > Rich
> > >
> > > 
> > > > Do you have loaded ztdummy as module?
> > > >
> > > >
> > > > -Mensaje original-
> > > > Trying to test the music on hold function and can't seem to get it to
> > work.
> > > > If anyone has it running, could you give me a clue? (I have googled and
> > > > found lots of questions, but no real suggestions.)
> > > >
> > > > I downloaded and installed the mpg123 package. From the RH9 console I
> > can
> > > > start the executable and hear the music via the speakers. The executable
> > > > is located in /usr/bin. (That works!)
> > > >
> > > > I set the following line in musiconhold.conf file:
> > > >   default => mp3:/var/lib/asterisk/mohmp3/
> > > > and when I do a ps ax |grep mpg123 I get:
> > > >   15798 pts/0S  0:00 /usr/bin/mpg123 -q -s --mono -r 8000 -b
> > 2048
> > > > For-You.mp3
> > > > If I stop asterisk, the above line disappears. (That works!)
> > > >
> > > > For lab work, the sip.conf has:
> > > >  allow=ulaw
> > > > only.
> > > >
> > > > In the zapata.conf file I have:
> > > >   musiconhold=default
> > > >
> > > > On asterisk startup, the console window shows:
> > > >   == Parsing '/etc/asterisk/musiconhold.conf': Found
> > > >   WARNING[1074504864]: File res_musiconhold.c, Line 506 (moh_register):
> > > > Unable to
> > > >   open pseudo channel for timing...  Sound may be choppy.
> > > >   == Registered application 'MusicOnHold'
> > > >   == Registered application 'WaitMusicOnHold'
> > > >   == Registered application 'SetMusicOnHold'
> > > > which looks like its okay.
> > > >
> > > > The the asterisk command line, I see:
> > > > -- Called 3002
> > > > -- SIP/3002-3ed3 is ringing
> > > > -- SIP/3002-3ed3 is ringing
> > > > -- SIP/3002-3ed3 answered SIP/3000-0e7b
> > > > -- Attempting native bridge of SIP/3000-0e7b and SIP/3002-3ed3
> > > > -- Started music on hold, class 'default', on SIP/3002-3ed3
> > > > but nothing is heard on either C7960's or Snom 200.
> > > >
> > > > I have not modified anything in sip.conf or extensions.conf; am I
> > suppose
> > > > to?
> > > >
> > > > What am I missing
> > > >
> > > > Rich
> > > >
> > > >
> > > >
> > > > ___
> > > > Asterisk-Users mailing list
> > > > [EMAIL PROTECTED]
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > > > ___
> > > > Asterisk-Users mailing list
> > > > [EMAIL PROTECTED]
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > > ---End of Original Message-
> > >
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> ---End of Original Message-
> 
> 
> ___
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> http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
BTEL Consulting
850-484-4535 x2111 (Office)
504-595-3916 x2111 (Experimental)
877-552-0838 (Backup Phone)

___

Re: [Asterisk-Users] help on MOH config, pretty close?

[Rich Adamson]

After following your suggestions, I get

[EMAIL PROTECTED] zaptel]# /sbin/modprobe ztdummy
/lib/modules/2.4.20-18.9/misc/zaptel.o: kernel-module version mismatch
/lib/modules/2.4.20-18.9/misc/zaptel.o was compiled for kernel version 2
.4.20-6
while this kernel is version 2.4.20-18.9.
/lib/modules/2.4.20-18.9/misc/zaptel.o: insmod /lib/modules/2.4.20-18.9/misc/zap
tel.o failed
/lib/modules/2.4.20-18.9/misc/zaptel.o: insmod ztdummy failed
[EMAIL PROTECTED] zaptel]# 

I should have two X100P Digium cards arriving tomorrow. Rather then mess with this,
should I just wait until those cards are installed?



> If I'm not mistaken, ztdummy makes work moh and meetme...
> Do lsmod from sh. If you don't see ztdummy, download zaptel, and modify
> Makefile. Line 89 must be:
> 
> MODULES=zaptel.o tor2.o torisa.o wcusb.o wcfxo.o wcfxs.o \
> ztdynamic.o ztd-eth.o wct1xxp.o wct4xxp.o ztdummy.o
> 
> (Remove # preceding ztdummy.o)
> 
> Then, do a make and make install. When finish, do:
> 
> modprobe ztdummy
> 
> Meetme and MOH should work now.
> 
> 
> - Original Message -
> From: "Rich Adamson" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Tuesday, September 09, 2003 9:59 PM
> Subject: RE: [Asterisk-Users] help on MOH config, pretty close?
> 
> 
> > The show modules command line suggests that is Not loaded. Should if be,
> and
> > if so, where do I do that?
> >
> > Rich
> >
> > 
> > > Do you have loaded ztdummy as module?
> > >
> > >
> > > -Mensaje original-
> > > Trying to test the music on hold function and can't seem to get it to
> work.
> > > If anyone has it running, could you give me a clue? (I have googled and
> > > found lots of questions, but no real suggestions.)
> > >
> > > I downloaded and installed the mpg123 package. From the RH9 console I
> can
> > > start the executable and hear the music via the speakers. The executable
> > > is located in /usr/bin. (That works!)
> > >
> > > I set the following line in musiconhold.conf file:
> > >   default => mp3:/var/lib/asterisk/mohmp3/
> > > and when I do a ps ax |grep mpg123 I get:
> > >   15798 pts/0S  0:00 /usr/bin/mpg123 -q -s --mono -r 8000 -b
> 2048
> > > For-You.mp3
> > > If I stop asterisk, the above line disappears. (That works!)
> > >
> > > For lab work, the sip.conf has:
> > >  allow=ulaw
> > > only.
> > >
> > > In the zapata.conf file I have:
> > >   musiconhold=default
> > >
> > > On asterisk startup, the console window shows:
> > >   == Parsing '/etc/asterisk/musiconhold.conf': Found
> > >   WARNING[1074504864]: File res_musiconhold.c, Line 506 (moh_register):
> > > Unable to
> > >   open pseudo channel for timing...  Sound may be choppy.
> > >   == Registered application 'MusicOnHold'
> > >   == Registered application 'WaitMusicOnHold'
> > >   == Registered application 'SetMusicOnHold'
> > > which looks like its okay.
> > >
> > > The the asterisk command line, I see:
> > > -- Called 3002
> > > -- SIP/3002-3ed3 is ringing
> > > -- SIP/3002-3ed3 is ringing
> > > -- SIP/3002-3ed3 answered SIP/3000-0e7b
> > > -- Attempting native bridge of SIP/3000-0e7b and SIP/3002-3ed3
> > > -- Started music on hold, class 'default', on SIP/3002-3ed3
> > > but nothing is heard on either C7960's or Snom 200.
> > >
> > > I have not modified anything in sip.conf or extensions.conf; am I
> suppose
> > > to?
> > >
> > > What am I missing
> > >
> > > Rich
> > >
> > >
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > ---End of Original Message-
> >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users

---End of Original Message-


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Re: [Asterisk-Users] help on MOH config, pretty close?

[CW_ASN]

If I'm not mistaken, ztdummy makes work moh and meetme...
Do lsmod from sh. If you don't see ztdummy, download zaptel, and modify
Makefile. Line 89 must be:

MODULES=zaptel.o tor2.o torisa.o wcusb.o wcfxo.o wcfxs.o \
ztdynamic.o ztd-eth.o wct1xxp.o wct4xxp.o ztdummy.o

(Remove # preceding ztdummy.o)

Then, do a make and make install. When finish, do:

modprobe ztdummy

Meetme and MOH should work now.


- Original Message -
From: "Rich Adamson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, September 09, 2003 9:59 PM
Subject: RE: [Asterisk-Users] help on MOH config, pretty close?


> The show modules command line suggests that is Not loaded. Should if be,
and
> if so, where do I do that?
>
> Rich
>
> 
> > Do you have loaded ztdummy as module?
> >
> >
> > -Mensaje original-
> > Trying to test the music on hold function and can't seem to get it to
work.
> > If anyone has it running, could you give me a clue? (I have googled and
> > found lots of questions, but no real suggestions.)
> >
> > I downloaded and installed the mpg123 package. From the RH9 console I
can
> > start the executable and hear the music via the speakers. The executable
> > is located in /usr/bin. (That works!)
> >
> > I set the following line in musiconhold.conf file:
> >   default => mp3:/var/lib/asterisk/mohmp3/
> > and when I do a ps ax |grep mpg123 I get:
> >   15798 pts/0S  0:00 /usr/bin/mpg123 -q -s --mono -r 8000 -b
2048
> > For-You.mp3
> > If I stop asterisk, the above line disappears. (That works!)
> >
> > For lab work, the sip.conf has:
> >  allow=ulaw
> > only.
> >
> > In the zapata.conf file I have:
> >   musiconhold=default
> >
> > On asterisk startup, the console window shows:
> >   == Parsing '/etc/asterisk/musiconhold.conf': Found
> >   WARNING[1074504864]: File res_musiconhold.c, Line 506 (moh_register):
> > Unable to
> >   open pseudo channel for timing...  Sound may be choppy.
> >   == Registered application 'MusicOnHold'
> >   == Registered application 'WaitMusicOnHold'
> >   == Registered application 'SetMusicOnHold'
> > which looks like its okay.
> >
> > The the asterisk command line, I see:
> > -- Called 3002
> > -- SIP/3002-3ed3 is ringing
> > -- SIP/3002-3ed3 is ringing
> > -- SIP/3002-3ed3 answered SIP/3000-0e7b
> > -- Attempting native bridge of SIP/3000-0e7b and SIP/3002-3ed3
> > -- Started music on hold, class 'default', on SIP/3002-3ed3
> > but nothing is heard on either C7960's or Snom 200.
> >
> > I have not modified anything in sip.conf or extensions.conf; am I
suppose
> > to?
> >
> > What am I missing
> >
> > Rich
> >
> >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ---End of Original Message-
>
>
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Re: [Asterisk-Users] freebsd and asterisk ?? anyone yet

[Chris Albertson]

I agree 100%. Sometimes code works on one system
because of a "quirk".  Building and testing on
multiple systems and debugging the autoconf scripts
has a way of making the code mature and robust.  I
keep and old DEC Alpha (a 64 bit machine) and SPARC
(big endian byte order) and run Solaris and Linux. 
You'd be amazed at the kinds of errors testing on
multiple platforms turns up.

> Once the code gets up to spec to be compiled on BSD
> systems, most likely
> it won't need much after that. When something is
> written with a 'linuxism'
> that breaks it from compiling on other platforms,
> its probably something
> that should get fixed anyways (for sanity's sake).
> Chances are it'll
> compile on a lot of other systems as well, heck,
> I'll build it on Tru64 to
> see if it works :)
> 
> > The other aspect of this whole thing, is that most
> people's PBX is an
> > entirely standalone machine (or should be).  Aside
> from AGI/etc which
> > generally have to run locally (but are also
> usually very cross-platform
> > anway), I can't understand why someone would
> prefer it on one OS over
> > another.
> 
> Aside from stability, security, and updatability,
> there's no particular
> reason. My network is almost entirely FreeBSD, to
> have a Linux box in the
> mix is a pain, especially when it comes to updating
> the dang thing.
> 
> > Does it really matter what OS it runs on at this
> point, as long as it's
> > a robust kernel?
> 
> I have yet to experience a really robust Linux
> kernel :)
> Doesn't mean they don't exist though... I think.
> 
> > Feel free to prove me long -- generally I'm a
> "right OS for the job" kind
> > of guy, but in the case of "appliances", it seems
> like the above logic
> > makes sense.
> 
> I've seen a lot of appliances built on bad
> platforms. Giving people the
> choice of OS only makes the SW platform stronger.
> 
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>
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Re: [Asterisk-Users] SIP LD carrier

[Ernest W. Lessenger]

At 07:57 PM 9/9/2003 -0400, you wrote:
Travis Johnson wrote:
> I've called NuFone and was not impressed by their voicemail answering 
system (choppy) and was unable to even leave a message before the phone 
call was disconnected (in the middle of the
>recording).

So your going to judge our system by making one phone call into my home
asterisk system that runs on a fully saturated ADSL connection.
One must take first impressions from something, and if that happens to be 
your answering machine...

Typical,
Unfortunately, yes.

Seriously folks, NuFone has very reasonable prices, easy setup and decent 
tech support (or so I've heard - I haven't had any problems with them so 
far). I urge you to give NuFone a try, if only because (one of?) their big 
cheeses reads the * mailing list :) I recently checked out the competition, 
and here's what I found:

NuFone: cool first impression, good second impression, good implementation
Deltathree: good first impression, poor second impression
Net2Phone: good first impression, excellent second impression, cool third 
impression
Vonage: cool first impression, cool second impression

In all cases I was evaluating for price, compatibility with * and ease of 
use/ease of billing. Of course, I had specific uses in minds, specific 
hardware, and my own company's cost structure in mind, so your mileage may 
vary.

--Ernest 

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RE: [Asterisk-Users] help on MOH config, pretty close?

[Rich Adamson]

The show modules command line suggests that is Not loaded. Should if be, and
if so, where do I do that?

Rich


> Do you have loaded ztdummy as module?
> 
> 
> -Mensaje original-
> Trying to test the music on hold function and can't seem to get it to work.
> If anyone has it running, could you give me a clue? (I have googled and
> found lots of questions, but no real suggestions.)
> 
> I downloaded and installed the mpg123 package. From the RH9 console I can
> start the executable and hear the music via the speakers. The executable
> is located in /usr/bin. (That works!)
> 
> I set the following line in musiconhold.conf file:
>   default => mp3:/var/lib/asterisk/mohmp3/
> and when I do a ps ax |grep mpg123 I get:
>   15798 pts/0S  0:00 /usr/bin/mpg123 -q -s --mono -r 8000 -b 2048
> For-You.mp3
> If I stop asterisk, the above line disappears. (That works!)
> 
> For lab work, the sip.conf has:
>  allow=ulaw
> only.
> 
> In the zapata.conf file I have:
>   musiconhold=default
> 
> On asterisk startup, the console window shows:
>   == Parsing '/etc/asterisk/musiconhold.conf': Found
>   WARNING[1074504864]: File res_musiconhold.c, Line 506 (moh_register):
> Unable to
>   open pseudo channel for timing...  Sound may be choppy.
>   == Registered application 'MusicOnHold'
>   == Registered application 'WaitMusicOnHold'
>   == Registered application 'SetMusicOnHold'
> which looks like its okay.
> 
> The the asterisk command line, I see:
> -- Called 3002
> -- SIP/3002-3ed3 is ringing
> -- SIP/3002-3ed3 is ringing
> -- SIP/3002-3ed3 answered SIP/3000-0e7b
> -- Attempting native bridge of SIP/3000-0e7b and SIP/3002-3ed3
> -- Started music on hold, class 'default', on SIP/3002-3ed3
> but nothing is heard on either C7960's or Snom 200.
> 
> I have not modified anything in sip.conf or extensions.conf; am I suppose
> to?
> 
> What am I missing
> 
> Rich
> 
> 
> 
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Re: [Asterisk-Users] SIP LD carrier

[Jeremy McNamara]

Travis Johnson wrote:

I've called NuFone and was not impressed by their voicemail answering system (choppy) and was unable to even leave a message before the phone call was disconnected (in the middle of the
recording).
 

So your going to judge our system by making one phone call into my home 
asterisk system that runs on a fully saturated ADSL connection.

Typical,

Jeremy McNamara



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RE: [Asterisk-Users] help on MOH config, pretty close?

[CW_ASN]

Do you have loaded ztdummy as module?


-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Rich Adamson
Enviado el: Martes, 09 de Septiembre de 2003 09:17 p.m.
Para: Asterisk-users-list
Asunto: [Asterisk-Users] help on MOH config, pretty close?



Trying to test the music on hold function and can't seem to get it to work.
If anyone has it running, could you give me a clue? (I have googled and
found lots of questions, but no real suggestions.)

I downloaded and installed the mpg123 package. From the RH9 console I can
start the executable and hear the music via the speakers. The executable
is located in /usr/bin. (That works!)

I set the following line in musiconhold.conf file:
  default => mp3:/var/lib/asterisk/mohmp3/
and when I do a ps ax |grep mpg123 I get:
  15798 pts/0S  0:00 /usr/bin/mpg123 -q -s --mono -r 8000 -b 2048
For-You.mp3
If I stop asterisk, the above line disappears. (That works!)

For lab work, the sip.conf has:
 allow=ulaw
only.

In the zapata.conf file I have:
  musiconhold=default

On asterisk startup, the console window shows:
  == Parsing '/etc/asterisk/musiconhold.conf': Found
  WARNING[1074504864]: File res_musiconhold.c, Line 506 (moh_register):
Unable to
  open pseudo channel for timing...  Sound may be choppy.
  == Registered application 'MusicOnHold'
  == Registered application 'WaitMusicOnHold'
  == Registered application 'SetMusicOnHold'
which looks like its okay.

The the asterisk command line, I see:
-- Called 3002
-- SIP/3002-3ed3 is ringing
-- SIP/3002-3ed3 is ringing
-- SIP/3002-3ed3 answered SIP/3000-0e7b
-- Attempting native bridge of SIP/3000-0e7b and SIP/3002-3ed3
-- Started music on hold, class 'default', on SIP/3002-3ed3
but nothing is heard on either C7960's or Snom 200.

I have not modified anything in sip.conf or extensions.conf; am I suppose
to?

What am I missing

Rich



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Re: [Asterisk-Users] freebsd and asterisk ?? anyone yet

[denon]

With regards to Asterisk on FBSD, I for one would love to see it happen.  I 
prefer FreeBSD over Linux in almost every case.

However, personally I have a few concerns:

Namely, the primary developer is a Linux nut .. (sorry Mark, I mean that in 
a good way :).  I'd hate to see development efforts get split up, and more 
time spent on porting/etc efforts, detracting from primary development.  If 
it's now slowing down new development, it's always a step behind while 
someone patches up the current builds.

The other aspect of this whole thing, is that most people's PBX is an 
entirely standalone machine (or should be).  Aside from AGI/etc which 
generally have to run locally (but are also usually very cross-platform 
anway), I can't understand why someone would prefer it on one OS over 
another.  I mean, by the time you strip down the linux install to make it 
rock-stable, it's basically a kernel, compiler, and a handful of 
dependances. Does it really matter what OS it runs on at this point, as 
long as it's a robust kernel?

Feel free to prove me long -- generally I'm a "right OS for the job" kind 
of guy, but in the case of "appliances", it seems like the above logic 
makes sense.

Thoughts?

-d

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[Asterisk-Users] help on MOH config, pretty close?

[Rich Adamson]

Trying to test the music on hold function and can't seem to get it to work.
If anyone has it running, could you give me a clue? (I have googled and
found lots of questions, but no real suggestions.)

I downloaded and installed the mpg123 package. From the RH9 console I can
start the executable and hear the music via the speakers. The executable
is located in /usr/bin. (That works!)

I set the following line in musiconhold.conf file:
  default => mp3:/var/lib/asterisk/mohmp3/
and when I do a ps ax |grep mpg123 I get:
  15798 pts/0S  0:00 /usr/bin/mpg123 -q -s --mono -r 8000 -b 2048 For-You.mp3
If I stop asterisk, the above line disappears. (That works!)

For lab work, the sip.conf has:
 allow=ulaw
only.

In the zapata.conf file I have:
  musiconhold=default 

On asterisk startup, the console window shows:
  == Parsing '/etc/asterisk/musiconhold.conf': Found
  WARNING[1074504864]: File res_musiconhold.c, Line 506 (moh_register): Unable to
  open pseudo channel for timing...  Sound may be choppy.
  == Registered application 'MusicOnHold'
  == Registered application 'WaitMusicOnHold'
  == Registered application 'SetMusicOnHold'
which looks like its okay.

The the asterisk command line, I see:
-- Called 3002
-- SIP/3002-3ed3 is ringing
-- SIP/3002-3ed3 is ringing
-- SIP/3002-3ed3 answered SIP/3000-0e7b
-- Attempting native bridge of SIP/3000-0e7b and SIP/3002-3ed3
-- Started music on hold, class 'default', on SIP/3002-3ed3
but nothing is heard on either C7960's or Snom 200.

I have not modified anything in sip.conf or extensions.conf; am I suppose to?

What am I missing

Rich



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Re: [Asterisk-Users] freebsd and asterisk ?? anyone yet

[Tom (UnitedLayer)]

On Tue, 9 Sep 2003, denon wrote:
> I'd hate to see development efforts get split up, and more time spent on
> porting/etc efforts, detracting from primary development.  If it's now
> slowing down new development, it's always a step behind while someone
> patches up the current builds.

Once the code gets up to spec to be compiled on BSD systems, most likely
it won't need much after that. When something is written with a 'linuxism'
that breaks it from compiling on other platforms, its probably something
that should get fixed anyways (for sanity's sake). Chances are it'll
compile on a lot of other systems as well, heck, I'll build it on Tru64 to
see if it works :)

> The other aspect of this whole thing, is that most people's PBX is an
> entirely standalone machine (or should be).  Aside from AGI/etc which
> generally have to run locally (but are also usually very cross-platform
> anway), I can't understand why someone would prefer it on one OS over
> another.

Aside from stability, security, and updatability, there's no particular
reason. My network is almost entirely FreeBSD, to have a Linux box in the
mix is a pain, especially when it comes to updating the dang thing.

> Does it really matter what OS it runs on at this point, as long as it's
> a robust kernel?

I have yet to experience a really robust Linux kernel :)
Doesn't mean they don't exist though... I think.

> Feel free to prove me long -- generally I'm a "right OS for the job" kind
> of guy, but in the case of "appliances", it seems like the above logic
> makes sense.

I've seen a lot of appliances built on bad platforms. Giving people the
choice of OS only makes the SW platform stronger.

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Re: [Asterisk-Users] Pushing data to a 7960

[David C. Troy]

Yeah, the only reason I ask is that pushing a URL is apparently possible
with the Call Manager firmware.  See Cisco::IPPhone perl module.  Was
hoping similar concept was possible for the SIP firmware... so if anybody
knows...

Dave

=
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ToadNet - Want to go fast?410-544-1329 FAX
570 Ritchie Highway, Severna Park, MD 21146-2925  www.toad.net

On Tue, 9 Sep 2003, Jared Smith wrote:

> On Tue, 2003-09-09 at 16:15, David C. Troy wrote:
> > Anybody have any experience pushing XML display data to a Cisco 7960?  I 
> > am pulling data from a webserver now, but I have stuff, like queue status 
> > info, I'd love to be able to push at particular phones.
> > 
> > A "pick up the damn phone" push message would be cool too. :)
> > 
> > Dave
> 
> As cool as that would be, I don't think the SIP firmware on the Ciscos
> supports "pushing" data to the phone.
> 
> (I'd love for someone out there to tell me I'm wrong!)
> 
> Jared
> 
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Re: [Asterisk-Users] Pushing data to a 7960

[Jared Smith]

On Tue, 2003-09-09 at 16:15, David C. Troy wrote:
> Anybody have any experience pushing XML display data to a Cisco 7960?  I 
> am pulling data from a webserver now, but I have stuff, like queue status 
> info, I'd love to be able to push at particular phones.
> 
> A "pick up the damn phone" push message would be cool too. :)
> 
> Dave

As cool as that would be, I don't think the SIP firmware on the Ciscos
supports "pushing" data to the phone.

(I'd love for someone out there to tell me I'm wrong!)

Jared

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Re: [Asterisk-Users] Endpoint-to-Endpoint RTP Packets

[andy]

Hi all,

I'm interested in using asterisk WITHOUT codec support: I work in a LAN, with no 
bandwidth, delay, ... problems; I use a Cisco GW as PSTN interface and when I 
use asterisk the overall delay is to high and the quality drops.

In particular, I'm interested in using asterisk as h323 to sip translator (and 
viceversa).
do you have any suggestion?
thanks

Andres


Quoting Mike Ciholas <[EMAIL PROTECTED]>:

> 
> On Tue, 9 Sep 2003, Eric Wieling wrote:
> 
> > Transcoding would be required for access to ANY of the asterisk
> > sound files, voicemail and PSTN via Zap interfaces.
> 
> If you are using G711 ulaw from the SIP phones, and that is what
> you are getting from the T1 PSTN link, would * have to transcode
> that?  Is there more to it than digital to digital copy?  Perhaps 
> echo canceling?
> 
> Can we also store sound files in ulaw?  I know that takes more 
> space, but perhaps it is less CPU work to move the bits around 
> than to codec them.
> 
> -- 
> Mike Ciholas(812) 476-2721 voice
> CIHOLAS Enterprises (812) 476-2881 fax
> 2626 Kotter Ave, Unit D [EMAIL PROTECTED]
> Evansville, IN 47715http://www.ciholas.com
> 
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> 
> 




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Re: [Asterisk-Users] delay problem in h323

[andy]

yes, I agree with you.
I verify with a sniffer and asterisk manages RTP flows. The problem is asterisk 
decode and then code again RTP flows. This function requires 5-7% CPU On my 
test-box (Linux rh 7.3 on P3 600 GHz). This solution  don't scale without dedicated 
HW, I think!

Another problem is codec supported: ok for G.711, G.729. I don't know for GSM 
BUT: what about video codec? what about proprietary codec or ciphered codec?

Do you have any suggestion on how I can manage this with asterisk? I'm very 
interested into asterisk as sip-to-h323 translator.
Thanks

Andrea


Quoting Steven Thomas <[EMAIL PROTECTED]>:

> 
> 
> 
> 
> 
> The only way I was able to solve my delay issue with Chan_oh323 was to
> switch to Chan_h323.
> 
> Chan_oh323 caused a similar 3 -4 sec delay on one way of the conversation.
> Checking the CPU stats on asterisk during the call - confirms that the RTP
> stream was somehow routing through asterisk - not sure why!
> 
> 
> 
> Regards,
> 
> Steven Thomas
> 
> 
> 
> 
>  
>
>   andrea <[EMAIL PROTECTED]> 
>
>   Sent by:  To:  
> [EMAIL PROTECTED]
>   [EMAIL PROTECTED]cc:  
>
>   .digium.com   Subject:  Re:
> [Asterisk-Users] delay problem in h323 
>  
>
>  
>
>   10-09-03 12:45 AM  
>
>   Please respond to  
>
>   asterisk-users 
>
>  
>
> 
> 
> 
> Hi all,
> 
> is it possible to disable RTP routing through asterisk? RTP routing is a
> very nice feature but, I think it’s important also to disable it in some
> cases (e. g. in a LAN).
> Do you have any suggestion?
> 
> Andrea
> 
> Rattana BIV wrote:
> 
> > Hi,
> >
> > I have a delay between two H323.
> >
> > Netmeeting1 - ||
> >  | gnuGK | --- [asterisk-oh323]
> > | Asterisk |
> > Netmeeting2 --||
> >
> > Netmmeting 1 call Netmeeting 2. When Netmmeting 1 speak Netmeeting 2
> > receive the voice without delay. But in the other way I have 3 secondes
> > delay.
> > In oh323.conf  I set jittermin and jittermax to 20, the ipTos=lowdelay.
> > I try to find where I can delete the delay.
> > Does anyone have a tip ?
> >
> >
> > Best Regards
> > Rattana
> >
> 
> 
> 
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Re: [Asterisk-Users] BudgeTone-100 Early Dial

[Steve Meyers]

On Tue, 2003-09-09 at 16:03, Tilghman Lesher wrote:
> Why not just use DISA:
> 
> exten => 9,1,DISA(no-password|outgoing)

Because I didn't know about it. :)  I'll try it out.

Steve

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[Asterisk-Users] UK Caller ID and X100p

[Robert Boardman]

Hi
I really need caller id to work in the UK, I understand that the X100p 
uses a US chipset,two questions
1) is that a product that converts UK to US caller id in line

or

2) would it be possible to have modem that supports CID  in parallel 
with the line and the x100p.The modem reads the line and reports the 
cid  to asterisk

Just thinking outloud

Robb

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[Asterisk-Users] Pushing data to a 7960

[David C. Troy]

Anybody have any experience pushing XML display data to a Cisco 7960?  I 
am pulling data from a webserver now, but I have stuff, like queue status 
info, I'd love to be able to push at particular phones.

A "pick up the damn phone" push message would be cool too. :)

Dave

=
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ToadNet - Want to go fast?410-544-1329 FAX
570 Ritchie Highway, Severna Park, MD 21146-2925  www.toad.net

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Re: [Asterisk-Users] delay problem in h323

[Steven Thomas]

The only way I was able to solve my delay issue with Chan_oh323 was to
switch to Chan_h323.

Chan_oh323 caused a similar 3 -4 sec delay on one way of the conversation.
Checking the CPU stats on asterisk during the call - confirms that the RTP
stream was somehow routing through asterisk - not sure why!



Regards,

Steven Thomas




   
  
  andrea <[EMAIL PROTECTED]>   
  
  Sent by:  To:   [EMAIL PROTECTED]

  [EMAIL PROTECTED]cc: 
 
  .digium.com   Subject:  Re: [Asterisk-Users] 
delay problem in h323 
   
  
   
  
  10-09-03 12:45 AM
  
  Please respond to
  
  asterisk-users   
  
   
  



Hi all,

is it possible to disable RTP routing through asterisk? RTP routing is a
very nice feature but, I think it’s important also to disable it in some
cases (e. g. in a LAN).
Do you have any suggestion?

Andrea

Rattana BIV wrote:

> Hi,
>
> I have a delay between two H323.
>
> Netmeeting1 - ||
>  | gnuGK | --- [asterisk-oh323]
> | Asterisk |
> Netmeeting2 --||
>
> Netmmeting 1 call Netmeeting 2. When Netmmeting 1 speak Netmeeting 2
> receive the voice without delay. But in the other way I have 3 secondes
> delay.
> In oh323.conf  I set jittermin and jittermax to 20, the ipTos=lowdelay.
> I try to find where I can delete the delay.
> Does anyone have a tip ?
>
>
> Best Regards
> Rattana
>



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Re: [Asterisk-Users] BudgeTone-100 Early Dial

[Tilghman Lesher]

On Tuesday 09 September 2003 16:46, Steve Meyers wrote:
> On Tue, 2003-09-09 at 11:41, Hielke Christian Braun wrote:
> > I have one problem with the BudgeTone phones and early dial. When i
> > dial a long external number with 9+, * starts to dial to early with
> > just a few digits. The outgoing call is placed through the SIP
> > provider Nikotel. Is there some timeout i can increase so that *
> > waits for all the digits before placing the SIP call? The firmware
> > on the phones is 1.0.3.81 and they use SIP Info to sent DTMF.
> > Sending via inband or RFC2833 did not work at all. The * version is
> > a week old from CVS. When not using early dial it works fine.
>
> I told the Grandstream guys about the problem about a month ago, they
> said they'd look into it.  The BudgeTones handle 4-5 digits okay (I
> can't remember which), but at some point they crap out from too many
> 484's.
>
> The way I handled it was to make the extension 9 go to a context that
> plays a fake dialtone in the background, and handles the actual phone
> number from there.  In my main context, I have:
>
> exten => 9,1,Goto(dialtone,s,1)



Why not just use DISA:

exten => 9,1,DISA(no-password|outgoing)

-Tilghman

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[Asterisk-Users] * Picks up line during outgoing call

[Matt Lawson]

We have some regular POTS phones connected to our incoming line as well 
as the machine that runs Asterisk.  Sometimes during an outgoing call 
from the POTS phone, the Asterisk will pick up also, and play its menu. 
The FXO card is set to fxs_ks signalling; I'm told this might be the 
culprit but I really don't understand about the signalling types and 
what the ramifications of different ones are.

Any help?  Thx.



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RE: [Asterisk-Users] Is there a web interface to the asterisk system?

[Paul Crick]

> It should be very easy to do, generate config files (or meta
> config files) from web gui.
Do you mean like storing config information in a separate database, then
generating the plaintext config files everytime there's a change? This is
very similar to what a lot of the web hosting control panels do, and works
well provided there's a way to let users include any custom/bespoke stuff
they need to (perhaps with an include file etc). This gives you the nice
control of a web based interface without being so rigid it becomes
inflexible or controlling.

> Another thing is that a configuration interface is what makes
> you product seem superior to others. So if people make good
> gui's for *, the probably keep them to themselves as a
> competetive advantage.
You think? Nah, we're all a friendly bunch here, no? It'd go against the
spirit of things if someone developed a kick ass web admin interface then
said "I'm alright Jack, I'm keeping this for myself". As the British would
say: "It's just not cricket!"

That said, I'd love something that let me define regular extensions and
devices through a web based interface. The problem comes with everyone
having different requirements. I want to do the whole voicemail macro thing,
you want a busy tone and nothing else.

I guess there's something to be said for the technical expertise of hacking
away at text config files in vi - suits me fine :-)

Cheers
Paul

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Re: [Asterisk-Users] Caller ID

[Tais M. Hansen]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On Tuesday 09 September 2003 17:54, Tais M. Hansen wrote:
> How do I optionally hide the caller id on outgoing calls on chan_zap? Ie.
> calling h323 -> asterisk -> chan_zap -> isdn provider.

Problem solved. I made app_dial.c take an option to change 
hidecallerid/restrictcid flag.

- -- 
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.2 (GNU/Linux)

iD8DBQE/XkvX2TEAILET3McRAmIQAJ9Lk6A5r6CPBDlIYgcdb+XmI1UHXQCfdANP
WlQBNDX6qIIhzu14U3eAUy8=
=kmtq
-END PGP SIGNATURE-

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RE: [Asterisk-Users] ADSI phones?

[Paul Crick]

I've got an Aastra 390 (from Telus in BC, Canada) and have had a successful
ADSI script download. The Comedian Mail graphical interface works on it too.
I had a couple of glitches where voice prompts were being played but the
phone wasn't responsive but these aren't repeatable. I'd grab a phone, play
with it, see how it goes for you.

Cheers
Paul

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Re: [Asterisk-Users] BudgeTone-100 Early Dial

[Steve Meyers]

On Tue, 2003-09-09 at 11:41, Hielke Christian Braun wrote:
> I have one problem with the BudgeTone phones and early dial. When i
> dial a long external number with 9+, * starts to dial to early with
> just a few digits. The outgoing call is placed through the SIP provider
> Nikotel. Is there some timeout i can increase so that * waits 
> for all the digits before placing the SIP call? The firmware on the 
> phones is 1.0.3.81 and they use SIP Info to sent DTMF. Sending via 
> inband or RFC2833 did not work at all. The * version is a week old
> from CVS. When not using early dial it works fine. 

I told the Grandstream guys about the problem about a month ago, they
said they'd look into it.  The BudgeTones handle 4-5 digits okay (I
can't remember which), but at some point they crap out from too many
484's.

The way I handled it was to make the extension 9 go to a context that
plays a fake dialtone in the background, and handles the actual phone
number from there.  In my main context, I have:

exten => 9,1,Goto(dialtone,s,1)

Then I have a dialtone context:

[dialtone]
exten => s,1,Answer
exten => s,2,Background(dialtone)
exten => 11,1,Macro(localcall-number,911)
exten => 911,1,Macro(localcall)
exten => _NXX,1,Macro(localcall)
exten => _1NXXNXX,1,Macro(localcall)

And a couple Macros:

[macro-localcall]
exten => s,1,Macro(localcall-number,${MACRO_EXTEN})
   
 [macro-localcall-number]
;${ARG1} - number to call
exten => s,1,Dial(${POTSGROUP}/${ARG1})
exten => s,2,Playback(vm-goodbye)
exten => s,3,Hangup
exten => s,102,Congestion

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Re: [Asterisk-Users] Is there a web interface to the asterisk system?

[mawali]

I have asked this question in the start, but then realized that this 
qustion does not generate any response at all. Going through the archive I 
find out that lot of people have claimed to have made web gui, but when I 
see them, I stick to vi.

It should be very easy to do, generate config files (or meta config files) 
from web gui. I think the webmin modules that you may have seen on the 
site is a failed (incomplete ??) attempt to do so.

Another thing is that a configuration interface is what makes you product 
seem superior to others. So if people make good gui's for *, the probably 
keep them to themselves as a competetive advantage.

Regards
FT


 On Tue, 9 Sep 2003, Buddy Edwards wrote:

> Is there a basic web interface to the console to the asterisk system
> like webmin?
> 

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Re: [Asterisk-Users] BudgeTone-100 Early Dial

[WipeOut .]

> Hello,
> 
> this is my first post to the list. I have to say that i am really
> impressed with *. Every day i find a new great feature to play
> around with. Also the SIP support has come a long way in just half a
> year. Just wanted to thank the programmers for this great software.
> 
I Agree!!

> 
> I have one problem with the BudgeTone phones and early dial. When i
> dial a long external number with 9+, * starts to dial to early with
> just a few digits. The outgoing call is placed through the SIP provider
> Nikotel. Is there some timeout i can increase so that * waits 
> for all the digits before placing the SIP call? The firmware on the 
> phones is 1.0.3.81 and they use SIP Info to sent DTMF. Sending via 
> inband or RFC2833 did not work at all. The * version is a week old
> from CVS. When not using early dial it works fine. 
> 
Then don't use early dial.. so you have to wait 5 seconds when you make a call.. so 
what.. or you can dial the number and hit the send button if you really don't want to 
wait the 5 seconds..

Later..
-- 
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Now with e-mail forwarding for only US$5.95/yr

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RE: [Asterisk-Users] ADSI phones?

[Wade J. Weppler]

There was a ton of discussion on the Aastra's.  Please check the list
archives:

http://lists.digium.com

> -Original Message-
> From: Ken Godee [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, September 09, 2003 4:54 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] ADSI phones?
> 
> Any suggestions? The Aastra 480 & 390 seem popular along
> with the CybioLink.
> Does anyone use these phones (or others)?
> 
> Are they compatible with atsterisk's ADSI?
> If so, how are people programming these phones?
> Searched thru archives, lots of previous talk but
> no soild info. I'd like to get a couple of ADSI phones
> to play with, just hate to waste the money if they won't
> work with *.
> 
> TIA
> 
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[Asterisk-Users] Is there a web interface to the asterisk system?

[Buddy Edwards]

Is there a basic web interface to the console to the asterisk
system like webmin?

Re: [Asterisk-Users] Endpoint-to-Endpoint RTP Packets

[Eric Wieling]

It would have to do some kind of trascoding, but it's a non-issue since
G729 is not involved and the CPU overhead is minimal.


On Tue, 2003-09-09 at 15:26, Mike Ciholas wrote:
> On Tue, 9 Sep 2003, Eric Wieling wrote:
> 
> > Transcoding would be required for access to ANY of the asterisk
> > sound files, voicemail and PSTN via Zap interfaces.
> 
> If you are using G711 ulaw from the SIP phones, and that is what
> you are getting from the T1 PSTN link, would * have to transcode
> that?  Is there more to it than digital to digital copy?  Perhaps 
> echo canceling?
> 
> Can we also store sound files in ulaw?  I know that takes more 
> space, but perhaps it is less CPU work to move the bits around 
> than to codec them.
-- 
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504-595-3916 x2111 (Experimental)
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[Asterisk-Users] ADSI phones?

[Ken Godee]

Any suggestions? The Aastra 480 & 390 seem popular along
with the CybioLink.
Does anyone use these phones (or others)?
Are they compatible with atsterisk's ADSI?
If so, how are people programming these phones?
Searched thru archives, lots of previous talk but
no soild info. I'd like to get a couple of ADSI phones
to play with, just hate to waste the money if they won't
work with *.
TIA

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[Asterisk-Users] SIP LD carrier

[Travis Johnson]

Hi,

We are installing a new Asterisk PBX system and need to find a VoIP carrier
that will handle all our long-distance services and tie in directly with the
Asterisk server. Any recommendations? I've called NuFone and was not impressed
by their voicemail answering system (choppy) and was unable to even leave a
message before the phone call was disconnected (in the middle of the
recording).

Thanks,

Travis
Microserv
 

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RE: [Asterisk-Users] Request for comments on queue statistics

[Dave Weis]

On Tue, 9 Sep 2003, Paul Crick wrote:
> > I have done some trival work with matrix orbital lcd
> > to show some stats counts, calls parked etc Just find
> > lcd a bit small do you have lead on bigger LED signs
> > that you have used b4 ??
> I've used a Beta-Brite sign which is pretty similar to a ProLite in
> functionality, just made by a different company. They're on eBay all the
> time, search for "LED sign" as well as the two brand names and you're bound
> to find something.

If you need something bigger try www.translux.com

dave

-- 
Dave Weis "I believe there are more instances of the abridgment
[EMAIL PROTECTED]   of the freedom of the people by gradual and silent
  encroachments of those in power than by violent 
  and sudden usurpations."- James Madison

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RE: [Asterisk-Users] Request for comments on queue statistics

[Paul Crick]

> I have done some trival work with matrix orbital lcd
> to show some stats counts, calls parked etc Just find
> lcd a bit small do you have lead on bigger LED signs
> that you have used b4 ??
I've used a Beta-Brite sign which is pretty similar to a ProLite in
functionality, just made by a different company. They're on eBay all the
time, search for "LED sign" as well as the two brand names and you're bound
to find something.

Cheers
Paul

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Re: [Asterisk-Users] Asterisk Security vulnerability report

[Steven Critchfield]

On Tue, 2003-09-09 at 15:23, Brian Jones wrote:
> I heard about this a while ago too.  How come I didn't hear anything about
> it from asterisk-announce?  (at least I don't recall receiving any emails
> about it.
> 
> Also, is there any plans in the future to create a stable and development
> branches of code?  Upgrading to the lastest CVS version may be difficult for
> some who have complex installations.  It would be easier just to recieve a
> patch for the stable version.

Depending on your installation, you may just want to turn off SIP
support to anything not currently supporting SIP. All you need to do is
add a noload=chan_sip.so to the modules.conf file. This is what I did to
our main switch as it has no need for SIP support. SIP can be handled by
a non critical machine and linked to the critical PSTN switch via IAX if
I need it to.   


> - Original Message - 
> From: "Lubomir Christov" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>; <[EMAIL PROTECTED]>
> Sent: Tuesday, September 09, 2003 3:54 PM
> Subject: [Asterisk-Users] Asterisk Security vulnerability report
> 
> 
> > Hello,
> >
> > today I found this security report regarding Asterisk SIP Security.
> >
> > http://www.securiteam.com/securitynews/5LP0720B5G.html
> >
> > Maybe It could help somebody who isn't using a newer than 15th of August
> > cvs version.
> >
> > Best regards
> > Lubo
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> 
> 
> ___
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-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Endpoint-to-Endpoint RTP Packets

[Mike Ciholas]

On Tue, 9 Sep 2003, Eric Wieling wrote:

> Transcoding would be required for access to ANY of the asterisk
> sound files, voicemail and PSTN via Zap interfaces.

If you are using G711 ulaw from the SIP phones, and that is what
you are getting from the T1 PSTN link, would * have to transcode
that?  Is there more to it than digital to digital copy?  Perhaps 
echo canceling?

Can we also store sound files in ulaw?  I know that takes more 
space, but perhaps it is less CPU work to move the bits around 
than to codec them.

-- 
Mike Ciholas(812) 476-2721 voice
CIHOLAS Enterprises (812) 476-2881 fax
2626 Kotter Ave, Unit D [EMAIL PROTECTED]
Evansville, IN 47715http://www.ciholas.com

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Re: [Asterisk-Users] Asterisk Security vulnerability report

[Brian Jones]

I heard about this a while ago too.  How come I didn't hear anything about
it from asterisk-announce?  (at least I don't recall receiving any emails
about it.

Also, is there any plans in the future to create a stable and development
branches of code?  Upgrading to the lastest CVS version may be difficult for
some who have complex installations.  It would be easier just to recieve a
patch for the stable version.

Brian.



- Original Message - 
From: "Lubomir Christov" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; <[EMAIL PROTECTED]>
Sent: Tuesday, September 09, 2003 3:54 PM
Subject: [Asterisk-Users] Asterisk Security vulnerability report


> Hello,
>
> today I found this security report regarding Asterisk SIP Security.
>
> http://www.securiteam.com/securitynews/5LP0720B5G.html
>
> Maybe It could help somebody who isn't using a newer than 15th of August
> cvs version.
>
> Best regards
> Lubo
>
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>
>


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RE: [Asterisk-Users] Channelbanks

[David Carr]

For FXS only, I like the CAC Access Bank 1s just because of the price. You
can get 24 FXS ports for around $200 from ebay or resellers.

For FXO or FXS/FXO, I like the CAC Adit 600s. They support FXO callerID
(which the AB1s don't), have much better disconnect supervision than the
AB1s, and they have 3 8-port cards instead of 2 12-port cards. That way you
can do 8 FXO + 16 FXS or many other combinations. You can get 24 FXS ports
for $500 - $600 and then buy additional FXO cards for $200 each or FXS cards
for $75 each. One chassis actually supports 6 8-port cards with 2 T1s (48
channels). It also allows you to break out one T1 into some voice channels
and some data channels which is nice for smaller offices. It also is
configurable via telnet (ethernet) which I never cared about until I had it
and now I wouldn't go back.

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Troy Settle
> Sent: Monday, September 08, 2003 11:34 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Channelbanks
>
>
>
> Ok, the Zhone sucks and the Adtran 750/850 seems to be a little too
> expensive.
>
> Can anyone recommend a decent channelbank that won't break the bank?
>
> TIA,
>
> --
>   Troy Settle
>   Pulaski Networks
>   http://www.psknet.com
>   540.994.4254 ~ 866.477.5638
>   Pulaski Chamber 2002 Small Business Of The Year
>
>
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Re: [Asterisk-Users] Endpoint-to-Endpoint RTP Packets

[Eric Wieling]

Codecs are always an issue.  Best to put disallow=all and
allow=whatevercodecyouwant in each [sipuser] entry.  You can't have
Asterisk do codec translation (transcoding) bewteen g729 and some other
codec unless you have the g729 licenses (US$10/channel from Digium).

Transcoding would be required for access to ANY of the asterisk sound
files, voicemail and PSTN via Zap interfaces.

On Tue, 2003-09-09 at 14:51, Ernest W. Lessenger wrote:
> At 02:38 PM 9/9/2003 -0500, you wrote:
> >That would be reinvite= and canreinvite= in the user entry for each SIP
> >endpoint.  Asterisk will allow the endpoints to talk directly to each
> >other if both those settings are = yes (the default, I think) AND both
> >endpoints use the same protocol (SIP) AND the same codec.
> 
> So Asterisk will allow it... and if I set both to no, asterisk would act as 
> a true proxy, using the most bandwidth efficient codec available for each 
> leg of the call (i.e. GSM for x-lite and g.729 for Cisco et al)?
> 
> Thanks,
> --Ernest
> 
> 
> >On Tue, 2003-09-09 at 13:04, Sean P. Robertson wrote:
> > > I have seen this asked in the archives several times, but do not see a
> > > definitive answer anywhere. Is there a way to tell the Asterisk to act like
> > > a "normal" SIP Proxy, handling only the SIP messages, and letting the 
> > RTP go
> > > point-to-point?
> > > - Original Message -
> > > From: "Sean Figgins" <[EMAIL PROTECTED]>
> > > To: <[EMAIL PROTECTED]>
> > > Sent: Tuesday, September 09, 2003 1:40 PM
> > > Subject: Re: [Asterisk-Users] freebsd and asterisk ?? anyone yet
> > >
> > >
> > > > On Mon, 8 Sep 2003, Jim Mercer wrote:
> > > >
> > > > > > Can we bribe you? :)
> > > > >
> > > > > sure, pay my rent for 3 months and give me a 50" plasma TV to play in
> > > the
> > > > > background.
> > > >
> > > > Is that all?  That sounds rather cheap, compared to the things direction
> > > > that I'd have to go if I wanted to stick to the cisci CM route, with
> > > > licenses for every endpoint that I want to connect.
> > > >
> > > > Realistically...  I just can not comprehend how to get stuff to work
> > > > correctly with Linux.  I used to be a Linux nut years ago, but once I
> > > > found FreeBSD with it's ports collection, I wondered why anyone ever
> > > > bothered with Linux and it's completely messed up software install
> > > > requirements.
> > > >
> > > > Right now, under RedHat 9.0, I have * running, but no hardware, and I
> > > > can't figure out how to get h.323 operational so I can talk to my cisco
> > > > gateway with the PRI interface...  I'm only guessing that FreeBSD 
> > would be
> > > > much easier for non-programmers like myself.
> > > >
> > > > -Sean
> > > >
> > > > ___
> > > > Asterisk-Users mailing list
> > > > [EMAIL PROTECTED]
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > >
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> >--
> >BTEL Consulting
> >850-484-4535 x2111 (Office)
> >504-595-3916 x2111 (Experimental)
> >877-552-0838 (Backup Phone)
> >
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> 
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Re: [Asterisk-Users] Endpoint-to-Endpoint RTP Packets

[Sean P. Robertson]

Thank you.  I saw references to these two settings in various places but did
not see how both were used together.

I appreciate it.

Sean

- Original Message -
From: "Eric Wieling" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, September 09, 2003 3:38 PM
Subject: Re: [Asterisk-Users] Endpoint-to-Endpoint RTP Packets


> That would be reinvite= and canreinvite= in the user entry for each SIP
> endpoint.  Asterisk will allow the endpoints to talk directly to each
> other if both those settings are = yes (the default, I think) AND both
> endpoints use the same protocol (SIP) AND the same codec.
>
> On Tue, 2003-09-09 at 13:04, Sean P. Robertson wrote:
> > I have seen this asked in the archives several times, but do not see a
> > definitive answer anywhere. Is there a way to tell the Asterisk to act
like
> > a "normal" SIP Proxy, handling only the SIP messages, and letting the
RTP go
> > point-to-point?



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Re: [Asterisk-Users] OT: Creating documentation using a web interface

[Chris Hirsch]

Heheh..lets try that again...we use TWiki http://twiki.org at work..and
I'm in the process of setting it up at home..its a GREAT way to
document anything. It starts as chaos and then eventually builds up a
well defined structure as you go along.

Chris

Chris Hirsch wrote:

  
  
You betcha!! We use it at work and we actually have non-html people
contributing and fixing typos and errors..its easy to set up and easy
to use.
  
Leif Madsen wrote:
  

  Hm

http://www.interactivetools.com/products/docbuilder/

This looks kind of what I want, but I am looking for a free version of
something preferably.  I would pay for something, but at this time I


am
  

  unable to because of commitments to having to pay for tuition (still


in
  

  school).

Getting closer though. .. back to the google :)



Jac Kersing mentioned to me to try twiki.  Has anyone else used this
with success to create documentation?

Installing now as a test platform.

Thanks,
Leif Madsen.

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[Asterisk-Users] Asterisk Security vulnerability report

[Lubomir Christov]

Hello,

today I found this security report regarding Asterisk SIP Security.

http://www.securiteam.com/securitynews/5LP0720B5G.html

Maybe It could help somebody who isn't using a newer than 15th of August 
cvs version.

Best regards
Lubo
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Re: [Asterisk-Users] Channelbanks

[TC]

>These are 2551 32ms echo cans. Each does 1-T1 and sits between the T100P 
>and the cb.
>Yes, the zapata echo canceller is turned off.
>And, the echo is gone. I can't even detect the training time.
CL
>Software is obviously a much more tidy approach, but they were cheap, 
>echo was a problem and I thought playing with them would help me better 
>understand the issues. Sure did.
and does the canceller run all fxs & fxo channels or both ???

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Re: [Asterisk-Users] Request for comments on queue statistics

[TC]

>I've played with LED readerboad signs before and they're pretty easy to
>program up - Does anyone else on the list have any interest in making these
>ACD stats available in this way?
I have done some trival work with matrix orbital lcd to show some stats
counts, calls parked etc
Just find lcd a bit small do you have lead on bigger LED signs that
you have used b4 ??

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Re: [Asterisk-Users] Endpoint-to-Endpoint RTP Packets

[Ernest W. Lessenger]

At 02:38 PM 9/9/2003 -0500, you wrote:
That would be reinvite= and canreinvite= in the user entry for each SIP
endpoint.  Asterisk will allow the endpoints to talk directly to each
other if both those settings are = yes (the default, I think) AND both
endpoints use the same protocol (SIP) AND the same codec.
So Asterisk will allow it... and if I set both to no, asterisk would act as 
a true proxy, using the most bandwidth efficient codec available for each 
leg of the call (i.e. GSM for x-lite and g.729 for Cisco et al)?

Thanks,
--Ernest

On Tue, 2003-09-09 at 13:04, Sean P. Robertson wrote:
> I have seen this asked in the archives several times, but do not see a
> definitive answer anywhere. Is there a way to tell the Asterisk to act like
> a "normal" SIP Proxy, handling only the SIP messages, and letting the 
RTP go
> point-to-point?
> - Original Message -
> From: "Sean Figgins" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Tuesday, September 09, 2003 1:40 PM
> Subject: Re: [Asterisk-Users] freebsd and asterisk ?? anyone yet
>
>
> > On Mon, 8 Sep 2003, Jim Mercer wrote:
> >
> > > > Can we bribe you? :)
> > >
> > > sure, pay my rent for 3 months and give me a 50" plasma TV to play in
> the
> > > background.
> >
> > Is that all?  That sounds rather cheap, compared to the things direction
> > that I'd have to go if I wanted to stick to the cisci CM route, with
> > licenses for every endpoint that I want to connect.
> >
> > Realistically...  I just can not comprehend how to get stuff to work
> > correctly with Linux.  I used to be a Linux nut years ago, but once I
> > found FreeBSD with it's ports collection, I wondered why anyone ever
> > bothered with Linux and it's completely messed up software install
> > requirements.
> >
> > Right now, under RedHat 9.0, I have * running, but no hardware, and I
> > can't figure out how to get h.323 operational so I can talk to my cisco
> > gateway with the PRI interface...  I'm only guessing that FreeBSD 
would be
> > much easier for non-programmers like myself.
> >
> > -Sean
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> ___
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--
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504-595-3916 x2111 (Experimental)
877-552-0838 (Backup Phone)

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Re: [Asterisk-Users] Endpoint-to-Endpoint RTP Packets

[Eric Wieling]

That would be reinvite= and canreinvite= in the user entry for each SIP
endpoint.  Asterisk will allow the endpoints to talk directly to each
other if both those settings are = yes (the default, I think) AND both
endpoints use the same protocol (SIP) AND the same codec.

On Tue, 2003-09-09 at 13:04, Sean P. Robertson wrote:
> I have seen this asked in the archives several times, but do not see a
> definitive answer anywhere. Is there a way to tell the Asterisk to act like
> a "normal" SIP Proxy, handling only the SIP messages, and letting the RTP go
> point-to-point?
> - Original Message -
> From: "Sean Figgins" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Tuesday, September 09, 2003 1:40 PM
> Subject: Re: [Asterisk-Users] freebsd and asterisk ?? anyone yet
> 
> 
> > On Mon, 8 Sep 2003, Jim Mercer wrote:
> >
> > > > Can we bribe you? :)
> > >
> > > sure, pay my rent for 3 months and give me a 50" plasma TV to play in
> the
> > > background.
> >
> > Is that all?  That sounds rather cheap, compared to the things direction
> > that I'd have to go if I wanted to stick to the cisci CM route, with
> > licenses for every endpoint that I want to connect.
> >
> > Realistically...  I just can not comprehend how to get stuff to work
> > correctly with Linux.  I used to be a Linux nut years ago, but once I
> > found FreeBSD with it's ports collection, I wondered why anyone ever
> > bothered with Linux and it's completely messed up software install
> > requirements.
> >
> > Right now, under RedHat 9.0, I have * running, but no hardware, and I
> > can't figure out how to get h.323 operational so I can talk to my cisco
> > gateway with the PRI interface...  I'm only guessing that FreeBSD would be
> > much easier for non-programmers like myself.
> >
> > -Sean
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
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504-595-3916 x2111 (Experimental)
877-552-0838 (Backup Phone)

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Re: [Asterisk-Users] Endpoint-to-Endpoint RTP Packets

[Sean P. Robertson]

Does that mean that you do not know? :)

Sean
___

Sean Robertson

NETXUSA
p. 800-289-6389
f.  864-233-4344  "Ask me about Voice over IP."
http://www.netxusa.com/

- Original Message -
From: "Steven Critchfield" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, September 09, 2003 2:40 PM
Subject: Re: [Asterisk-Users] Endpoint-to-Endpoint RTP Packets


> I wonder if your company took your name off of the support contact page
> because you do not know how to properly use your email software yet.
>
> Can you justify your abuse of the reply button where you only changed
> the subject line and didn't even remove the non relevant message below?
>
> Stupid, 'tarded user.
>
>
> On Tue, 2003-09-09 at 13:04, Sean P. Robertson wrote:
> > I have seen this asked in the archives several times, but do not see a
> > definitive answer anywhere. Is there a way to tell the Asterisk to act
like
> > a "normal" SIP Proxy, handling only the SIP messages, and letting the
RTP go
> > point-to-point?
> >
> > Sean
> > ___
> >
> > Sean Robertson
> >
> > NETXUSA
> > p. 800-289-6389
> > f.  864-233-4344  "Ask me about Voice over IP."
> > http://www.netxusa.com/
> >



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RE: [Asterisk-Users] SIP Status Codes

[Adam Roach]

ftp://ftp.rfc-editor.org/in-notes/rfc3261.txt

The text you want is on pages 182 - 193.

/a

> -Original Message-
> From: Steven J. Sobol [mailto:[EMAIL PROTECTED]
> Sent: Monday, September 08, 2003 12:35
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] SIP Status Codes
> 
> 
> 
> Can anyone give me a pointer to descriptions of the status codes my
> Grandstream phone displays? I've looked on Google but can't find a
> definitive listing of SIP codes.
> 
> -- 
> JustThe.net Internet & Multimedia Services
> 22674 Motnocab Road * Apple Valley, CA 92307-1950 
> Steve Sobol, Proprietor 
> 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED]
> 
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> 
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[Asterisk-Users] Is t.38 fax relay supported in Asterisk?

[Lee Goodman]

I thought I saw a thread that said that a pseudo modem driver was built, but
only for h.323.
Could it also work for SIP?

Are there any details?

Thanks

Lee Goodman

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Re: [Asterisk-Users] Endpoint-to-Endpoint RTP Packets

[Steven Critchfield]

I wonder if your company took your name off of the support contact page
because you do not know how to properly use your email software yet. 

Can you justify your abuse of the reply button where you only changed
the subject line and didn't even remove the non relevant message below? 

Stupid, 'tarded user. 
 

On Tue, 2003-09-09 at 13:04, Sean P. Robertson wrote:
> I have seen this asked in the archives several times, but do not see a
> definitive answer anywhere. Is there a way to tell the Asterisk to act like
> a "normal" SIP Proxy, handling only the SIP messages, and letting the RTP go
> point-to-point?
> 
> Sean
> ___
> 
> Sean Robertson
> 
> NETXUSA
> p. 800-289-6389
> f.  864-233-4344  "Ask me about Voice over IP."
> http://www.netxusa.com/
> 
> - Original Message -
> From: "Sean Figgins" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Tuesday, September 09, 2003 1:40 PM
> Subject: Re: [Asterisk-Users] freebsd and asterisk ?? anyone yet
> 
> 
> > On Mon, 8 Sep 2003, Jim Mercer wrote:
> >
> > > > Can we bribe you? :)
> > >
> > > sure, pay my rent for 3 months and give me a 50" plasma TV to play in
> the
> > > background.
> >
> > Is that all?  That sounds rather cheap, compared to the things direction
> > that I'd have to go if I wanted to stick to the cisci CM route, with
> > licenses for every endpoint that I want to connect.
> >
> > Realistically...  I just can not comprehend how to get stuff to work
> > correctly with Linux.  I used to be a Linux nut years ago, but once I
> > found FreeBSD with it's ports collection, I wondered why anyone ever
> > bothered with Linux and it's completely messed up software install
> > requirements.
> >
> > Right now, under RedHat 9.0, I have * running, but no hardware, and I
> > can't figure out how to get h.323 operational so I can talk to my cisco
> > gateway with the PRI interface...  I'm only guessing that FreeBSD would be
> > much easier for non-programmers like myself.
> >
> > -Sean
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Request for comments on queue statistics

[TC]

>As a first step towards getting real queue statistics going (a project
>arguably adjunct to Asterisk itself), I propose that the Queue app be
>modified to produce its own "Queue Log," which could be stored either as
>text or in a SQL table.  For simplicity I would probably just implement a
>MySQL backend.


This is always a thorn, I would argue this should be done via a
database astraction layer, database like opinons, everybody has thier own

>The "Queue Log" could record events such as when an agent took a call,
>when a call was abandoned, when a call enters the queue, etc.
>
>It's been suggested that queue stats can be gleaned from CDR, but I submit
>that they really can't be, as there is no way to tell queue holdtime,
>whether a call is abandoned, etc.  In theory the Queue Log could link back
>into the CDR by way of the unique identifier field
AGREED this is most certainly the correct way to do it, then we can make
simple
joins to the the cdr & the queue table  via this foreign key relationship
>and serve as an enhancement to CDR; but it can exist be independent of CDR.
Let CDR do
>its thing, let the queue app worry about queue logging.
TOTALLY AGREED
>Once a rich Queue Log is available, it becomes fairly easy to generate the
>sorts of statistics people have been asking for.  Thoughts?
Yup this is the right high level approach we now need to agree on the
schema(s),
and the events of interest
To get the ball rolling & very incomplete, off the top of my head :)
I'd suggest with have a
queueEntry- the header table 1 for each unique call
 -QueueEntryID - PK for this table
 -UniqueCallID - FK to the CDR entry
 -QueueName
 -StartTime
 -WaitTime
 -Agent
 -Group
 ...kitchen sink
queueEntryEvent - many events that are recorded against a given unique
queueEntry
I guess i could also submit to a denormalized single table as well :)

queueEntryEvent - the header table 1 for each unique call
 -QueueEntryEventID - FK for this table
 -QueueEntryID - FK to table queueEntry
 -QueueEventType
 -StartTime
 -End Time
 -Note (arbitrary stuff that is unique to an event type)
...
Then in queue.conf
queueDB=csv, mysql, ..whatever
queueLog - yes/no to log calls on this queue
queueEvents -definintion of events of interest that will trigger entry into
the queueEntry Tables
  - that is defined for each queue
  -a comma delimited list of event names of interest
   eg join, leave, hungup, exitToContext, Duration, Parked,
Transfered ...





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Re: [Asterisk-Users] Xlite = no sound

[Peter Pauly]

On Tue, Sep 09, 2003 at 11:04:34AM +, WipeOut . wrote:
> Where did you get access to X-Ten.com's CVS server?
> 
> I didn't know they had the source code for x-lite available..
>

Sorry, I should have been more clear - I used the latest
version of Asterisk via CVS. 
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RE: [Asterisk-Users] Fax

[David Carr]

The hardware we are using for the asterisk server is a 2U server with dual
Xeon procs. It has a 4-port T1 card in it. Two ports are used for voice T1s.
Port #3 connects to an Adit 600 channel bank with 8 FXO ports and 16 FXS
ports. Our hylafax modems are plugged into the fxs ports. The modems we use
for hylafax are BestData external (serial) modems that support callerID
(~$35 each).

Although I've never tried to fax over sip, we have faxed successfully over
iax on a 100 Mbps LAN. We haven't had much luck trying to fax over iax to
remote offices on our wan, although I can't say we have tested it since Mark
has added his latest enhancements. Faxing over the wan would be an
improvement where physical machines are involved, but not needed for hylafax
where our best results have occured when the fax fxs ports are on the same
machine that receives the inbound call so asterisk just does a zap native
bridge.

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Troy Settle
> Sent: Tuesday, September 09, 2003 11:06 AM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Fax
>
>
>
> David,
>
> Could you elaborate on what hardware you're using for this?
>
> Also, speaking of faxes and *, I know that you can't fax over IAX(2),
> but it seems that faxing over SIP (ala ATA-186) works fine?  Would it be
> possible to set up the fax extension on one * box that can then use SIP
> to get the call to a second * box that's sitting ~10ms away?
>
>
> --
>   Troy Settle
>   Pulaski Networks
>   http://www.psknet.com
>   540.994.4254 ~ 866.477.5638
>   Pulaski Chamber 2002 Small Business Of The Year
>
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of David Carr
> > Sent: Monday, September 08, 2003 4:57 PM
> > To: [EMAIL PROTECTED]
> > Subject: RE: [Asterisk-Users] Fax
> >
> >
> > The way we do it is our T1 comes in with unlimited DIDs. In
> > our case we just
> > order more toll-free numbers, each with its own DNIS. Then we
> > have four FXS
> > ports (Zap/g2) connected to hylafax modems. When the call
> > comes in using
> > DNIS 1234, asterisk sets the callerID name to "1234", sends
> > the call to
> > Zap/g2, and our hylafax config routes the fax to email
> > [EMAIL PROTECTED] Then in our mail table we
> > forward each mail
> > alias to where we really want the fax to go.
> >
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> > > [mailto:[EMAIL PROTECTED] Behalf Of Ernest W.
> > > Lessenger
> > > Sent: Monday, September 08, 2003 12:21 PM
> > > To: [EMAIL PROTECTED]
> > > Subject: Re: [Asterisk-Users] Fax
> > >
> > >
> > > At 07:52 PM 9/8/2003 +0200, you wrote:
> > > >Is there a way to configure Hylafax or sth & one modem
> > behind an ATA-186
> > > >to email faxes to different adresses depending on the
> > called number ?
> > >
> > > I've looked into this myself, and I think the answer is
> > "yes, with some
> > > minor code changes." My thought is that you would use a
> > separate HylaFax
> > > server with six modems in it, and add two Digium FXS cards to the
> > > * server.
> > > Configure * to send the faxes out the correct FXS port for
> > each company,
> > > and configure hylafax to queue the faxes to a different
> > folder for each
> > > line. User interface and notification are left as an exercise to the
> > > reader, as is the actual hylafax configuration :)
> > >
> > > The major downside to the above is all the POTS lines you have to
> > > run, and
> > > the waste of ports. An alternative would be to use only one
> > or two POTS,
> > > and have * set the CallerID for each company. Then, have Hylafax
> > > queue the
> > > incoming faxes based on CallerID. The disadvantage to this
> > is, of course,
> > > that you lose any real CallerID information.
> > >
> > > --Ernest
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
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[Asterisk-Users] Endpoint-to-Endpoint RTP Packets

[Sean P. Robertson]

I have seen this asked in the archives several times, but do not see a
definitive answer anywhere. Is there a way to tell the Asterisk to act like
a "normal" SIP Proxy, handling only the SIP messages, and letting the RTP go
point-to-point?

Sean
___

Sean Robertson

NETXUSA
p. 800-289-6389
f.  864-233-4344  "Ask me about Voice over IP."
http://www.netxusa.com/

- Original Message -
From: "Sean Figgins" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, September 09, 2003 1:40 PM
Subject: Re: [Asterisk-Users] freebsd and asterisk ?? anyone yet


> On Mon, 8 Sep 2003, Jim Mercer wrote:
>
> > > Can we bribe you? :)
> >
> > sure, pay my rent for 3 months and give me a 50" plasma TV to play in
the
> > background.
>
> Is that all?  That sounds rather cheap, compared to the things direction
> that I'd have to go if I wanted to stick to the cisci CM route, with
> licenses for every endpoint that I want to connect.
>
> Realistically...  I just can not comprehend how to get stuff to work
> correctly with Linux.  I used to be a Linux nut years ago, but once I
> found FreeBSD with it's ports collection, I wondered why anyone ever
> bothered with Linux and it's completely messed up software install
> requirements.
>
> Right now, under RedHat 9.0, I have * running, but no hardware, and I
> can't figure out how to get h.323 operational so I can talk to my cisco
> gateway with the PRI interface...  I'm only guessing that FreeBSD would be
> much easier for non-programmers like myself.
>
> -Sean
>
> ___
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>


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Re: [Asterisk-Users] DBPut and DBGet performance

[Tilghman Lesher]

On Tuesday 09 September 2003 06:00, Surajee Ratnayake wrote:
> hi,
>
> This question is about DBPut and DBGet,
> Can i put about 1000  keys in a single family, (only once for the
> lifetime) for ex.
> exten => _X.,5,DBput(family/key1=${val})
> ...
> exten => _X.,5,DBput(family/key1000=${val})
>
> like above and if i later retrieve it, randomely, with inbound calls,
> will it affect performance?

Nope, it should work just fine.

-Tilghman

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Re: [Asterisk-Users] Channelbanks

[Steve Brown]

TC wrote:

Hey hey..
What are the model number of these Tellabls echo Cancellers
do you then turn off all asterisk echo Cancellation in zapata
Do you tuly have 0 echo now ???


 

These are 2551 32ms echo cans. Each does 1-T1 and sits between the T100P 
and the cb.
Yes, the zapata echo canceller is turned off.
And, the echo is gone. I can't even detect the training time.

Software is obviously a much more tidy approach, but they were cheap, 
echo was a problem and I thought playing with them would help me better 
understand the issues. Sure did.

Steve

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RE: [Asterisk-Users] Request for comments on queue statistics

[Paul Crick]

Hi David, group..

I definitely agree that some kind of external statistics would be of great
benefit. I implemented some event logging on an IVR system I worked on
recently, where events were stored in a table related back to the unique ID
in the CDR record (so one CDR, many events) and it worked quite well - nice
and easy to get answers to the questions management like to ask.

I also like the idea of logging to text files - it's plain and simple and
doesn't require the use of MySQL (not sure where we stand on licensing for
something like that?) (I guess not a big issue if this is an optional
thing?).

But bounce back to the other side of the fence, I'd have to say my
preference would be to have stats in a MySQL database purely for ease of
access. Have some pretty web pages displaying near real time queue
statistics etc.. sexy!

I guess one of the questions is whether you go with an event logging type
system, one call having multiple records, with the downside of having to
maybe do some other processing later (working out the time between events
etc?) or have a single record per call with multiple fields defining the
events of the call's lifetime. The record could only be written at the end
of the call which means no information would be available for calls in
progress, although maybe in-progress calls could have records written and
updated every xx seconds with a flag showing when the call/record is
complete? *shrugs* just tossing ideas around there..

I've played with LED readerboad signs before and they're pretty easy to
program up - Does anyone else on the list have any interest in making these
ACD stats available in this way?

Cheers
Paul

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Re: [Asterisk-Users] Callgroup, Pickupgroup and SIP

[James Sizemore]

know bug http://bugs.digium.com/bug_view_page.php?bug_id=116

WipeOut . wrote:

I have just started to play with callgroups and pickupgroups..

I updates my * from CVS this morning (about 15 mins ago)..

I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in sip.conf..

I place a call from phoneA to phoneB, then I go to phoneC and dial *8# , the call does not get picked up by phoneC and continues to ring on phoneB..

Have I not configured somthing correctly or is there a bug??

Later.
 



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Re: [Asterisk-Users] Callgroup, Pickupgroup and SIP

[James Sizemore]

Know bug
http://bugs.digium.com/bug_view_page.php?bug_id=116
Pertti Pikkarainen wrote:



I have problems with this as well ( similar config ).  My CVS is 10 
days old.

I can get the call picked up with *8 (  *8# does not work )  but 
the phone B never stops ringing.
B rings forever. I'm using SNOM200.

--Pertti

WipeOut . wrote:

I have just started to play with callgroups and pickupgroups..

I updates my * from CVS this morning (about 15 mins ago)..

I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone 
configurations in sip.conf..

I place a call from phoneA to phoneB, then I go to phoneC and dial 
*8# , the call does not get picked up by phoneC and continues to ring 
on phoneB..

Have I not configured somthing correctly or is there a bug??

Later.
 




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[Asterisk-Users] BudgeTone-100 Early Dial

[Hielke Christian Braun]

Hello,

this is my first post to the list. I have to say that i am really
impressed with *. Every day i find a new great feature to play
around with. Also the SIP support has come a long way in just half a
year. Just wanted to thank the programmers for this great software.


I have one problem with the BudgeTone phones and early dial. When i
dial a long external number with 9+, * starts to dial to early with
just a few digits. The outgoing call is placed through the SIP provider
Nikotel. Is there some timeout i can increase so that * waits 
for all the digits before placing the SIP call? The firmware on the 
phones is 1.0.3.81 and they use SIP Info to sent DTMF. Sending via 
inband or RFC2833 did not work at all. The * version is a week old
from CVS. When not using early dial it works fine. 


Thanks in advance,
 Christian.




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Re: [Asterisk-Users] freebsd and asterisk ?? anyone yet

[Sean Figgins]

On Mon, 8 Sep 2003, Jim Mercer wrote:

> > Can we bribe you? :)
>
> sure, pay my rent for 3 months and give me a 50" plasma TV to play in the
> background.

Is that all?  That sounds rather cheap, compared to the things direction
that I'd have to go if I wanted to stick to the cisci CM route, with
licenses for every endpoint that I want to connect.

Realistically...  I just can not comprehend how to get stuff to work
correctly with Linux.  I used to be a Linux nut years ago, but once I
found FreeBSD with it's ports collection, I wondered why anyone ever
bothered with Linux and it's completely messed up software install
requirements.

Right now, under RedHat 9.0, I have * running, but no hardware, and I
can't figure out how to get h.323 operational so I can talk to my cisco
gateway with the PRI interface...  I'm only guessing that FreeBSD would be
much easier for non-programmers like myself.

-Sean

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[Asterisk-Users] ISDN TA

[Robert Boardman]

I have an ISDN TA that has 2 POTS interfases (FXS), can these be used 
with asterisk?

Thanks in advance

Robb

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[Asterisk-Users] Rhino Channel Bank

[George Pajari]

FYI I asked them:

> Your website talks about configuring the Rhino channel bank
> as 24xFXS.
> Is it possible to mix FXO and FXS modules? What affect does
> that have on pricing?
They replied:

We will have FXO and the ability to mix both FXS & FXO within 60-90
days.  Our R&D department is testing this upgrade as we speak.  

Pricing of this FXS upgrade has not been determined yet.  However, I
believe the retail price will be near $1,500.
George, thank you again for your interest in the RHINO Channel Bank. 
I will keep your contact information and let you know the moment our
FXO box is ready!


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RE: [Asterisk-Users] Most Basic System

[Paul Crick]

You could do it with a couple of soft phones if you have enough PCs to go
around - I did this a little bit, just to play. Then I forked out for a
Cisco 7940 and ATA-186 off eBay so I had some real phones to play with..
then I forked out for the Dev Kit Lite (see - it gets addictive, this
Asterisk stuff!).

So now I have a box that can accept 1 phone line and 1 analog/POTS phone,
with a Cisco 7940 connected, and sometimes I play with X-Lite (from
www.xten.com) to test stuff further.

Hope this helps
Paul

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Re: [Asterisk-Users] OT: Creating documentation using a web interface

[Chris Hirsch]

You betcha!! We use it at work and we actually have non-html people
contributing and fixing typos and errors..its easy to set up and easy
to use.

Leif Madsen wrote:

  
Hm

http://www.interactivetools.com/products/docbuilder/

This looks kind of what I want, but I am looking for a free version of
something preferably.  I would pay for something, but at this time I

  
  am
  
  
unable to because of commitments to having to pay for tuition (still

  
  in
  
  
school).

Getting closer though. .. back to the google :)

  
  
Jac Kersing mentioned to me to try twiki.  Has anyone else used this
with success to create documentation?

Installing now as a test platform.

Thanks,
Leif Madsen.

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Re: [Asterisk-Users] Getting a local number abroad - Newbie question

[Tilghman Lesher]

On Tuesday 09 September 2003 07:17, Maron Kristófersson wrote:
> Hello!
>
> I have a staff member abroad and need to provide him with the ability
> to make local calls.  The features I need are:
>
> * Possibillity to make calls at local (Icelandic) charges from
> Ireland office.

Sure.  Just send the IAX call into a context which has the permission
to access outside FXO channels on the other side.  You could use
DISA on the remote system, to give "local dialtone" or you could simply
pass a dialled extension through into a Dial command.

> * Possibillity to call the local Icelandic number and reach the
> Ireland office.

Yep, same as above, except ring an inside channel, instead of going to
an outside channel.

For example:
; Local outside line
exten => _9.,1,Dial(Zap/${EXTEN:1})
; Remote line
exten => _8.,1,Dial(IAX/[EMAIL PROTECTED]/${EXTEN:1})

; On the other system
exten => _9.,1,Dial(Zap/${EXTEN:1})
exten => 101,1,Dial(Zap/9)

So you'd dial 92345 to dial 2345 locally, 892345 to dial 2345 on the
remote system, and dial 8101 to dial the remote channel 9 (possibly a
remote office extension).

> I'm also wondering if there is any isdn based solution since there is
> a possibillity of another staff member going abroad, and then I would
> like to have 2 numbers, 1 for each user.

See here:  http://ns1.jnetdns.de/jn/relaunch/asterisk/

> Is the phone on the user end software-based, ip-telephone or an
> analog telephone?

Could be any or all of the above.

-Tilghman

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RE: [Asterisk-Users] Fax

[Troy Settle]

David,

Could you elaborate on what hardware you're using for this?

Also, speaking of faxes and *, I know that you can't fax over IAX(2),
but it seems that faxing over SIP (ala ATA-186) works fine?  Would it be
possible to set up the fax extension on one * box that can then use SIP
to get the call to a second * box that's sitting ~10ms away?


--
  Troy Settle
  Pulaski Networks
  http://www.psknet.com
  540.994.4254 ~ 866.477.5638
  Pulaski Chamber 2002 Small Business Of The Year
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of David Carr
> Sent: Monday, September 08, 2003 4:57 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Fax
> 
> 
> The way we do it is our T1 comes in with unlimited DIDs. In 
> our case we just
> order more toll-free numbers, each with its own DNIS. Then we 
> have four FXS
> ports (Zap/g2) connected to hylafax modems. When the call 
> comes in using
> DNIS 1234, asterisk sets the callerID name to "1234", sends 
> the call to
> Zap/g2, and our hylafax config routes the fax to email
> [EMAIL PROTECTED] Then in our mail table we 
> forward each mail
> alias to where we really want the fax to go.
> 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] Behalf Of Ernest W.
> > Lessenger
> > Sent: Monday, September 08, 2003 12:21 PM
> > To: [EMAIL PROTECTED]
> > Subject: Re: [Asterisk-Users] Fax
> >
> >
> > At 07:52 PM 9/8/2003 +0200, you wrote:
> > >Is there a way to configure Hylafax or sth & one modem 
> behind an ATA-186
> > >to email faxes to different adresses depending on the 
> called number ?
> >
> > I've looked into this myself, and I think the answer is 
> "yes, with some
> > minor code changes." My thought is that you would use a 
> separate HylaFax
> > server with six modems in it, and add two Digium FXS cards to the
> > * server.
> > Configure * to send the faxes out the correct FXS port for 
> each company,
> > and configure hylafax to queue the faxes to a different 
> folder for each
> > line. User interface and notification are left as an exercise to the
> > reader, as is the actual hylafax configuration :)
> >
> > The major downside to the above is all the POTS lines you have to
> > run, and
> > the waste of ports. An alternative would be to use only one 
> or two POTS,
> > and have * set the CallerID for each company. Then, have Hylafax
> > queue the
> > incoming faxes based on CallerID. The disadvantage to this 
> is, of course,
> > that you lose any real CallerID information.
> >
> > --Ernest
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> ___
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[Asterisk-Users] Request for comments on queue statistics

[David C. Troy]

All,

I've been doing a lot of work on the Queue application and have some ideas
I wanted to get some feedback on.  Many people, myself included, have
stated their desire for more/better queue statistics.  We ought to be able 
to bring things up to a standard comparable to the established ACD apps.

While some basic metrics (completed/abandoned calls, which agents last
took a call, etc) can be kept in memory for real-time use, any numbers 
handled in this way are neither auditable or storable for management 
review.

As a first step towards getting real queue statistics going (a project 
arguably adjunct to Asterisk itself), I propose that the Queue app be 
modified to produce its own "Queue Log," which could be stored either as 
text or in a SQL table.  For simplicity I would probably just implement a 
MySQL backend.

The "Queue Log" could record events such as when an agent took a call, 
when a call was abandoned, when a call enters the queue, etc.

It's been suggested that queue stats can be gleaned from CDR, but I submit
that they really can't be, as there is no way to tell queue holdtime,
whether a call is abandoned, etc.  In theory the Queue Log could link back
into the CDR by way of the unique identifier field and serve as an
enhancement to CDR; but it can exist be independent of CDR.  Let CDR do 
its thing, let the queue app worry about queue logging.

Once a rich Queue Log is available, it becomes fairly easy to generate the
sorts of statistics people have been asking for.  Thoughts?

Mark, does this contradict any sort of master plan on the subject?

Regards,
Dave

=
David C. Troy   [EMAIL PROTECTED]   410-384-2500 Sales
ToadNet - Want to go fast?410-544-1329 FAX
570 Ritchie Highway, Severna Park, MD 21146-2925  www.toad.net

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Re: [Asterisk-Users] Has the "allow=all" function changed in sip.conf?

[Ernest W. Lessenger]

At 11:26 AM 9/9/2003 -0600, you wrote:
What is the general consciences for the "allow=all" statement?  Should it be
used, should it be specific towards those codecs supported, or removed?
My understanding is that you MUST have at least one allow and one deny, or 
none at all. Just having one or the other causes "problems."

--Ernest 

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RE: [Asterisk-Users] Has the "allow=all" function changed in sip.conf?

[Leif Madsen]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Rich Adamson
> Sent: Tuesday, September 09, 2003 1:26 PM
> To: Asterisk-users-list
> Subject: [Asterisk-Users] Has the "allow=all" function 
> changed in sip.conf?

> What is the general consciences for the "allow=all" 
> statement?  Should it be used, should it be specific towards 
> those codecs supported, or removed?

And... is this syntax allowed:

disallow=all
allow=gsm
allow=ulaw
allow=alaw


- --
Leif Madsen - Telecommunications Technology
Sheridan College - Trafalgar Campus
@: [EMAIL PROTECTED]
ICQ: 3445119FWD: 18924
 

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[Asterisk-Users] Has the "allow=all" function changed in sip.conf?

[Rich Adamson]

I had posted earlier asking about a Snom200 communicating with a C7960
and lots of noise in one direction. Turned out the problem was created
by me removing the allow=all statement in sip.conf. Someone had suggested
that statement is no longer needed, and using allow=ulaw, etc, had an issue
where one or more deny's had to be used as well.

By adding allow=ulaw in the sip.conf file, the Snom now correctly interacts
with the C7960. Both phones have their defaults set to the same, however the
v2.1l Snom code apparently has a problem negotiating that correctly.

What is the general consciences for the "allow=all" statement?  Should it be
used, should it be specific towards those codecs supported, or removed?



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[Asterisk-Users] Caller ID

[Tais M. Hansen]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi,

How do I optionally hide the caller id on outgoing calls on chan_zap? Ie. 
calling h323 -> asterisk -> chan_zap -> isdn provider.

Using setcallerid() to clear the callerid won't work since my provider 
requires a callerid. AFAIK one has to send something along with the callerid 
(set a flag in the call setup or similar) to indicate that the callerid is 
private.

Any pointers?

- -- 
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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=XCMO
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[Asterisk-Users] Snom polling..

[WipeOut .]

For anyone who is interested it appears that the nes Snom firmware now gets the phone 
to poll for updates approx every hour.. I was wondering why my ISDN line was being 
activated all through the night..

I see there is a setting now to control updates so I am going to set that to "Never 
update, do not load settings" to see if it stops the polling..

Anyway its just an FYI for anyone with dial on demand links or any other reason wht 
thay would not want this happeneing..
-- 
__
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Re: [Asterisk-Users] Call Time out Problem-Very Urgent!

[Martin Pycko]

> yes, i had 'callprogress=yes', and i commented it.
> now the time out is working.
>
> Thank you very much
>
> by disabling callprogress in an analog environment, does it affet the
> call disconnection?
IT does but you should only use it for analog channels ... and propably
only FXOs.

Martin

>
> Surajee
>
> - Original Message -
> From: "Martin Pycko" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Monday, September 08, 2003 10:32 PM
> Subject: Re: [Asterisk-Users] Call Time out Problem-Very Urgent!
>
>
> > Do you have callprogress=yes in zapata.conf ? If yes, then comment it out.
> > Also you could send some trace from the console including "pri debug span
> > "
> >
> > Martin
> >
> > On Mon, 8 Sep 2003, Surajee Ratnayake wrote:
> >
> > > Is it a problem with E1, bcos, when we dial a SIP extension from the
> same
> > > asterisk box
> > > it timeouts but not the Zap ones..
> > > We tried without |t, but it didn't work..
> > > still keeps on ringing forever...
> > > :-(
> > >
> > > Surajee
> > >
> > >
> > > - Original Message -
> > > From: <[EMAIL PROTECTED]>
> > > To: "Surajee Ratnayake" <[EMAIL PROTECTED]>
> > > Sent: Sunday, September 07, 2003 12:29 PM
> > > Subject: Re: [Asterisk-Users] Call Time out Problem-Very Urgent!
> > >
> > >
> > > > surajee:
> > > >
> > > > what happens if you remove the |t ? still no timeout ?
> > > >
> > > >  -wasim
> > > >
> > > > On Mon, 8 Sep 2003, Surajee Ratnayake wrote:
> > > >
> > > > > hi,
> > > > >
> > > > > I have a problem in call time out,
> > > > > An ISDN PRI E1 from PSTN and another ISDN PRI E1 from a
> > > > > Nortel PBX is conneted to my server.
> > > > > But when i do a Dialout(from both E1s)the calls do not timeout.
> > > > > For ex.
> > > > >  Dial(Zap/g4/123456|20|t)
> > > > >
> > > > > suppose other side is ringing and is not answering.
> > > > > even after 20 seconds, call doesn't get timeout
> > > > >
> > > > > pls gv me a solutions..
> > > > > its really urgent..
> > > > >
> > > > > Surajee
> > > > >
> > > >
> > > >
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
>
> ___
> Asterisk-Users mailing list
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>

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Re: [Asterisk-Users] Channelbanks

[TC]

>FWIW, I also bought a rack of Tellabs echo cancellers for $50 and have 
>one between the cb and the T100p. A little overkill, but very effective.

Hey hey..
What are the model number of these Tellabls echo Cancellers
do you then turn off all asterisk echo Cancellation in zapata
Do you tuly have 0 echo now ???



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Re: [Asterisk-Users] Callgroup, Pickupgroup and SIP

[Jared Smith]

I have the same problem with Cisco 7960s.  I have found however that if
you do the *8 from a Zap channel, then the called phone stops ringing. 
But if I do *8 from a SIP device, the called phone continues to ring and
ring for at least a minute.

I think Mark is working on fixing this (or at least was looking at it
yesterday evening).

Jared

On Tue, 2003-09-09 at 02:07, Pertti Pikkarainen wrote:
> I have problems with this as well ( similar config ).  My CVS is 10 days 
> old.
> 
> I can get the call picked up with *8 (  *8# does not work )  but 
>  the phone B never stops ringing.
> B rings forever. I'm using SNOM200.
> 
> --Pertti
> 
> 
> WipeOut . wrote:
> 
> >I have just started to play with callgroups and pickupgroups..
> >
> >I updates my * from CVS this morning (about 15 mins ago)..
> >
> >I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations 
> >in sip.conf..
> >
> >I place a call from phoneA to phoneB, then I go to phoneC and dial *8# , the call 
> >does not get picked up by phoneC and continues to ring on phoneB..
> >
> >Have I not configured somthing correctly or is there a bug??
> >
> >Later.
> >  
> >

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RE: [Asterisk-Users] Channelbanks

[Neal]

This one seems less expensive.  Does anyone have any experience with Rhino? 
http://www.channelbanks.com/
We don't have one yet, but probably will soon.
Regards,
Neal



Ok, the Zhone sucks and the Adtran 750/850 seems to be a little too
expensive.

Can anyone recommend a decent channelbank that won't break the bank?

TIA,

--
  Troy Settle
  Pulaski Networks
  http://www.psknet.com
  540.994.4254 ~ 866.477.5638
  Pulaski Chamber 2002 Small Business Of The Year
 

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RE: [Asterisk-Users] Dynamic SIP outbound usernames?

[Zac Sprackett]

Add dial plan entries like this in extensions.conf

[trunks-ld]
; long distance
exten => _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],20,Tr)
exten => _1NXXNXX,2,Congestion

add a sip entry like this in sip.conf

[mt-1204]
type=peer
host=172.20.16.7
mask=255.255.255.255
dtmfmode=inband
context=default
qualify=yes
canreinvite=no

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Alastair Maw
> Sent: Tuesday, September 09, 2003 5:27 AM
> To: * Users List
> Subject: [Asterisk-Users] Dynamic SIP outbound usernames?
> 
> 
> Hi,
> 
> I have * set up as a PSTN->VoIP gateway (with an E1 with multiple 
> numbers pointing to it).
> 
> I'd really like to be able to dial out to a SIP server like so:
> 
>exten => _X.,1,Dial(SIP/[EMAIL PROTECTED])
> 
> I.e. the remote SIP server receives a SIP INVITE with a "To:" header 
> containing the dialed number (e.g. [EMAIL PROTECTED]).
> 
> This is equivalent to having a hostname extension in sip.conf with a 
> dynamic username of ${DNID}.
> 
> How does one achieve this?
> 
> Likewise, it would be nice to be able to use gnophone to simulate calls 
> into the system, by pointing it at the * box and getting the dialed 
> number on that to route things in the same way.
> 
> Any ideas?
> 
> -- 
> Alastair Maw <[EMAIL PROTECTED]>
> MX Telecom - Systems Analyst
> http://www.mxtelecom.com
> 
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Re: [Asterisk-Users] delay problem in h323

[andrea]

Hi all,

is it possible to disable RTP routing through asterisk? RTP routing is a 
very nice feature but, I think it’s important also to disable it in some 
cases (e. g. in a LAN).
Do you have any suggestion?

Andrea

Rattana BIV wrote:

Hi,
 
I have a delay between two H323.
 
Netmeeting1 - ||
 | gnuGK | --- [asterisk-oh323] 
| Asterisk |
Netmeeting2 --||
 
Netmmeting 1 call Netmeeting 2. When Netmmeting 1 speak Netmeeting 2 
receive the voice without delay. But in the other way I have 3 secondes 
delay.
In oh323.conf  I set jittermin and jittermax to 20, the ipTos=lowdelay.
I try to find where I can delete the delay.
Does anyone have a tip ?
 
 
Best Regards
Rattana
 


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Re: [Asterisk-Users] Channelbanks

[Steve Brown]

Troy Settle wrote:

Ok, the Zhone sucks and the Adtran 750/850 seems to be a little too
expensive.
Can anyone recommend a decent channelbank that won't break the bank?

TIA,

 

I've bought several vintage Premisys CB's on ebay for $75 to $150 /ea.

They work fine with my T100p's. This is not strictly a commercial 
environment. I'm using them to interconnect the phones between my 
Seattle and Florida homes.

The FXS interfaces are plentiful, pass caller id and seem to do what 
they are supposed to do.
The FXO interfaces are really scarce. They also don't do callerid 
without a recent cpu card, per the Zhone site. And, I haven't found one yet.
They are easy to configure. Google found several pdf manuals.

They seem randomly listed as Premisys IMACS/600 or 800, AT&T Paradyne 
Acculink Access Controller, ADC ICX and others.

FWIW, I also bought a rack of Tellabs echo cancellers for $50 and have 
one between the cb and the T100p. A little overkill, but very effective.

Steve

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[Asterisk-Users] Asterisk @ SMAU

[Matteo Brancaleoni]

Hi all.
On 2 october will start SMAU, here in Italy , in Milano.
SMAU is the biggest IT (and computer related stuff) expo event 
that we have in italy.
I'll be @ SMAU from 2/10 to 6/10 , in the opensource area,
where my company will promote asterisk & digium hardware.
If anyone will attend the expo, drop me an email off line,
so will be able to meet at the expo and chat a bit ;)

Matteo.

-- 
Matteo Brancaleoni
Espia System Administrator
Email : [EMAIL PROTECTED]
Web   : http://www.espia.it
Phone : +39 02 70633354  - ext 911
IAX(2): [EMAIL PROTECTED] - ext 911
Iaxtel: 1-700-56-62458   - ext 911

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[Asterisk-Users] delay problem in h323

[Rattana BIV]

Hi,
 
I have a delay between two H323. 
 
Netmeeting1 - |    
        |
 | gnuGK 
| --- [asterisk-oh323] | Asterisk |
Netmeeting2 --|    
        |
 
Netmmeting 1 call Netmeeting 2. When Netmmeting 1 
speak Netmeeting 2 receive the voice without delay. But in the other way I have 
3 secondes delay.

In oh323.conf  I set jittermin and jittermax 
to 20, the ipTos=lowdelay.
I try to find where I can delete the 
delay.
Does anyone have a tip ?
 
 
Best Regards
Rattana