date:20060512

Hello,

Digium does not provide snmp support to monitor their
cards !

Anybody has tried Sangoma product A104 Quad T1/E1 or
others ?

Regards
harry








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Re: [Asterisk-Users] features.conf *1 Call Recording

2006-05-12 Thread Giridhar Reddy Bandi

did you include automon = *1 in your features.conf ?? it should be somthing like this [featuremap]automon = *1 --Giridhar Bandi 
On 5/12/06, Dave Morrow [EMAIL PROTECTED] wrote:
Thanks for the response.How would I change the DTMF transfer mode?David MorrowTechnical Systems LeadAutodata Solutions Company[EMAIL PROTECTED]
http://www.autodatasolutions.comTel: (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of the way! This message has originated from Autodata Solutions. The attached
material is the Confidential and Proprietary Information of AutodataSolutions. This email and any files transmitted with it are confidentialand intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] mailto:
[EMAIL PROTECTED]-Original Message-From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of FabioSent: Thursday, May 11, 2006 6:55 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] features.conf *1 Call Recordingif you ar using SIP clients, try changing DTMF transfer mode.For test use sip debugon your * console, then place a call and watch the output. In INFO or
rfc2833 mode you must see the codes like SIP messages. If you are usinginband transfer mode (DTMF codes aretransferred like sounds) you don'tsee the codes.Also, try adjusting featuredigittimeout in features.conf
:[general]featuredigittimeout = 2000 ; 2 secondsbecause the default 500ms is a very short time.Fabio Olaechea3Tech SRLCalle 48 Nro 632, Of. 67.La Plata, CP B1900AMZBuenos Aires, Argentina.
Tel. +54 221 445 0244 Ext. 301Fax. +54 221 445 0245www.trestech.com.ar-Mensaje original-De: 
[EMAIL PROTECTED][mailto:[EMAIL PROTECTED]]En nombre de Dave MorrowEnviado el: Miercoles, 10 de Mayo de 2006 02:48 
p.m.Para: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: RE: [Asterisk-Users] features.conf *1 Call RecordingOK. You lost me.David MorrowTechnical Systems LeadAutodata Solutions Company
[EMAIL PROTECTED]http://www.autodatasolutions.comTel: (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of the way! 
This message has originated from Autodata Solutions. The attachedmaterial is the Confidential and Proprietary Information of AutodataSolutions. This email and any files transmitted with it are confidential
and intended solely for the use of the individual or entity to whom theyare addressed. If you have received this email in error please deletethis message and notify the Autodata system administrator at
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]-Original Message-From: 
[EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of AlejandroVargasSent: Wednesday, May 10, 2006 10:29 AM
To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] features.conf *1 Call Recording2006/5/10, Dave Morrow [EMAIL PROTECTED]
: I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success.Is there a trick to this?May be a problem with the way you are sending the dialtones. Try sending
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Re: [Asterisk-Users] CentOS 4.x and ooh323

2006-05-12 Thread Richard Scobie




Bruce Reeves wrote:
I'm trying to add ooh323c to my asterisk 1.2.7.1 http://1.2.7.1 
install and did an svn update of asterisk-addons and followed the readme 
in asterisk-ooh323c and I get through the .configure with no errors. But 
make causes:


rpath /usr/local/lib -L./ooh323c/src -version-info 1:1:0  -lpthread
make: rpath: Command not found
make: [libchan_h323.la] Error 127 (ignored)


My previous mail mentioned that this had been posted recently on the 
list, however I was confusing it with the ooh323 list on Sourceforge.


Unfortunately Sourceforge seems to be in the middle of a meltdown with 
CVS access to projects broken and for over a week, no list mail is being 
archived there, and I have deleted the mails I received relevent to this.


So you may want to try on that list. This was the first post about it 
before archiving stopped.


http://sourceforge.net/mailarchive/forum.php?thread_id=10291962forum_id=43045


Regards,

Richard
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[Asterisk-Users] Sip domains, contexts and CHECKSIPDOMAIN

2006-05-12 Thread Chris Hastie


Hi

I'm struggling with setting up SIP domains.

If I specify a domain and a context in [general], that context overrides 
any set in type=peer blocks elsewhere. This results in incoming calls 
from PSTN gateways I use arriving in the wrong context.


If I don't specify a context (which the docs I've found suggest is 
valid), then I get:


2006-05-12 07:36:16 WARNING[95290]: chan_sip.c:12539 reload_config: 
Empty context specified at line 43 for domain 'domain.com'


and the domain does not appear when I do a sip show domains. It isn't 
recognised as local, CHECKSIPDOMAIN doesn't do what I want and calls I 
want are rejected.


If I specify autodomain=yes, then the IP address and canonical hostname 
of the box are added to the domain list, and sip show domains shows them 
with a context of (default). It would appear that for incoming calls 
from PSTN gateways at least this does what I want, in that the context 
specified in the type=peer block is the one used. However, I can find no 
way to add other domains to the list with this '(default)' context. I 
particularly want to add the domain name, rather than the host's FQDN, 
because my internal SIP clients are all configured to use this. At the 
moment, specifying any domain but not that means the clients can't 
register, specifying that domain with a context of 'incoming' means the 
internal clients can't make out bound calls, and using the context 
'outbound' has huge security implications.


I'd like to get sip domains working if only because I'd like to change 
the difficult to maintain


exten = s,3,GoToIf($[${SIPDOMAIN} : ${LOCALREGEX}]?4:20)

in my dial plan to something like

exten = s,3,GoToIf($[${CHECKSIPDOMAIN(${SIPDOMAIN})} = ]?4:20)

I'm using Asterisk 1.2.7.1 on FreeBSD 5.4.
Thanks
--
Chris Hastie
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[Asterisk-Users] SIP/NAT disconnection issue

2006-05-12 Thread Scott Bussinger

Help! I'm having an odd problem that I'm not seeing in any of the list 
archives and thought I'd ask and see if anyone can help.

I've got Asterisk behind a NAT and an SPA-841 SIP phone behind a different 
NAT. Everything works fine (incoming calls ring, outgoing calls work, audio 
in both directions, etc.) except for one thing -- when I hangup the SIP 
phone, Asterisk never disconnects the call. I've hooked up a syslog server 
and it appears the phone is sending a BYE command. However, running with SIP 
DEBUG on the server shows no SIP activity when the disconnect occurs. 
There's no indication on the firewall log that any packets were blocked for 
any reason (it's a separate linux box serving as a firewall/router).

Presumably I've got a NAT issue of some sort that's causing my problem --  
can anyone suggest any possibilities as to what's wrong and what I might do 
to solve it? 



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RE: [Asterisk-Users] MeetME Conferencing

2006-05-12 Thread Josh McAllister

Ok, the script below (meetme.agi) will prompt for a valid pin up to 3 times. If 
the pin matches one of the defined Admin pins, it will set the dialplan 
priority to 10 and exit, if User, sets to 20 and exits. Otherwise Hangs up.

In the case of admin, these MeetMe options are used:
a - Admin mode
A - Marked mode
c - Announce number of participants (optional of course)
s - Present Admin menu by pressing '*'
x - close conf when last marked user leaves.

In the case of user:
c s x are used as above, but we add:
w - wait until marked user enters. (Plays MoH until then)

The dialplan assumes you have a static pinless conference setup as conf #10.

extensions.conf:
exten = 5552323,1,Wait(1)
exten = 5552323,2,Answer()
exten = 5552323,3,AGI(meetme.agi)
exten = 5552323,4,NoOp(Invalid Pin)
exten = 5552323,5,Hangup()

exten = 5552323,10,NoOp(Admin Pin)
exten = 5552323,11,MeetMe(10,aAcsx)
exten = 5552323,12,Hangup()

exten = 5552323,20,NoOp(User Pin)
exten = 5552323,21,MeetMe(10,cswx)
exten = 5552323,22,Hangup()



The script of course requires the Asterisk::AGI module.

meetme.agi:

#!/usr/bin/perl
use Asterisk::AGI;
my $AGI = new Asterisk::AGI;
my $input = { %{$AGI-ReadParse()} };

#our $DEBUG = 1; 

my @UserPins = ('1','2');
my @AdminPins = ('9','8');

my $mode = collectPin($AGI,5);

$AGI-verbose(collectPin got '$mode') if $DEBUG;

if ($mode eq 'Admin') {
   $AGI-set_priority(10);
} elsif ($mode eq 'User') {
   $AGI-set_priority(20);
} else {
   $AGI-stream_file(goodbye,'');
   $AGI-hangup;
}

exit;

sub collectPin {
   my $AGI = shift;
   my $maxdigits = shift;

   my $tries = 0;

   #Three tries to select an existing pin.
   while ($tries  3) {
  $AGI-stream_file(please-try-again,'') if $tries  0;
  $tries++;
  my $pin = $AGI-get_data('enter-conf-pin-number', 1, $maxdigits);
  $AGI-verbose(Got PIN $pin.) if $DEBUG;
  next unless $pin  0;

  if ( grep(/^$pin$/, @AdminPins) ) {
 $AGI-stream_file(pin-number-accepted,'');
 return 'Admin';
  } elsif ( grep(/^$pin$/, @UserPins) ) {
 $AGI-stream_file(pin-number-accepted,'');
 return 'User';
  } else {
 $AGI-stream_file(conf-invalidpin,'');
  }
   }

   return undef;
}


What can I say, I was bored.

Enjoy,

Josh McAllister

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Thursday, May 11, 2006 10:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] MeetME Conferencing

Static configs for the conference rooms are not an issue.

The main goal is to allow the moderator to determine when the conference 
“starts” by having all participants hearing MOH until the moderator starts the 
interactive call with a  PIN known only to the moderator, and then allowing the 
moderator (and only the moderator) to kick out all users from the keypad when 
the call is over.

An additional benefit would be gained if authenticate() or realtime() app 
commands could be used against a mysql database for the participant and 
moderator pins so an app could be written easily to allow changing of the PINS 
in the database.


From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Austin
Sent: Thursday, May 11, 2006 10:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] MeetME Conferencing

I believe you can accomplish this with a well crafted dialplan.
 
If you did not have the restriction against out of tree modules, I would
recommend an app that strores the conference details in a database
and would allow just this kind of control.
 
Dan


From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Thursday, May 11, 2006 4:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] MeetME Conferencing
Not opposed to paying someone that can do it right ☺
 
As far as “coding” goes, you mean create the dialplan entries, not modify the 
meetme source, correct?
 
Our application requires that this can be done in 1.2 release, not trunk and 
not with an add-in that is not part of 1.2
 
If you have done it and would like to charge for you knowledge PM me, if you 
are willing to post a sample free of charge do it here for the benefit of all.
 

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
Sent: Thursday, May 11, 2006 5:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] MeetME Conferencing
 
Nope not asking too much.
 
What you are asking for is possible and not unique but you may have to pay for 
someone to code it for you.
 
 
Cheers,
 
Dean
 
 
 

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Thursday, 11 May 2006 6:56

[Asterisk-Users] Alarmreciver finally found ATA

2006-05-12 Thread Andrew Nowrot

Hi,I finally found an ATA which works really well with asterisk and its application alarmreceiver. Frankly it works just like the TDM card. It is Soundwin S800 series ATA.CheersAndrew 
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Re: [Asterisk-Users] TE110P on E1

2006-05-12 Thread Gareth Blades

For BT in the UK I use :-

zaptel.conf
loadzone = uk
defaultzone=uk
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31

zapata.conf
[trunkgroups]
[channels]
language=en
context=did
priindication = outofband
usecallerid=yes
cidsignalling=v23
usecallingpres=yes
sendcalleridafter=1
switchtype = euroisdn
pridialplan=unknown
signalling = pri_cpe
group = 1
channel = 1-15
channel = 17-31
overlap=yes


On Fri, 2006-05-12 at 08:34, Koen Van Impe wrote:
 Hi,
  
 I wonder if anyone is using Digium's TE110P card on an E1 connection.
 I have been try to, but so far it wasn't much of a success.
 It only works more or less in EuroISDN as PRI CPE.
 And even that config gives me some trouble with channel negotiation.
  
 My current config:
 zaptel.conf:
 span=1,1,0,ccs,hdb3
 bchan=1-15,17-31
 dchan=16
 loadzone=be
 defaultzone=be
  
 zapata.conf:
 [trunkgroups]
 trunkgroup = 1,16
 spanmap = 1,1,1
  
 [channels]
 context=incoming-pri
 switchtype=euroisdn
 pridialplan=national
 prilocaldialplan=national
 signalling=pri_cpe
 group=1
 channels = 1-15,17-31
  
 I have tried EuroISDN and QSIG in both NET and CPE, without much
 success.
 From traces of D-channel messaging, I think there's a problem with
 channel negotiation.
 The information element (IE) involved only shows 5 bytes coming from
 our PBX.
 But Asterisk (Zaptel) uses 6 bytes. From pri debug messages on
 Asterisk, I see that it adds a DS1 Identifier.
 I haven't seen that in other d-channel traces between other systems.
  
 Anyone with experience on this matter???
  
 Regards,
  
 Koen
 
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Re: [Asterisk-Users] asterisk management interface

2006-05-12 Thread Umair Bari

Hello,

Try http://www.freepbx.org, its written in PHP with mysql at the end, it also uses .conf files for configurations.

regards,

Umair bari
On 5/8/06, moona ather [EMAIL PROTECTED] wrote:
Hi,I have to make a web-based management interface of configuring asteriski wanted to know if it is as simple as reading the .conf files and searching
for the required section in the file and adding users etc. or there areother steps involved too?? As I have seen many such built codes on this siteand found lots of code... kindly tell me how complex it is and how many
other steps are involved in making this interface as i am new in this.Emmo._Don't just search. Find. Check out the new MSN Search!
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[Asterisk-Users] Echo cancel: chan_misdn vs bristuff? HFC card vs expensive card?

2006-05-12 Thread Cosmin Prund


Hello everyone.

I've got a HFC ISDN card that I'm using with chan_misdn and it basically 
behaves like crap. Echo is waaay worst then echo I get TDM400 card, 
sound is choppy (there other side is allays complaining about sound 
interruptions) and to top it all it detects fake DTMF's all the time.


Is this a chan_misdn problem or is it a card problem? I really need to 
get this fix and I need to know the way to go. I don't want to throw 
money at a better card if the card is not the issue but if that's the 
only solution, I'll need to order the card ASAP!

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Re: [Asterisk-Users] regarding freepbx

2006-05-12 Thread Umair Bari

freepbx has been improved since then, and I believe if you edit/add something in original asterisk .conf files, it stays there. I've tried it long ago when it was called AMP and it worked.

regards,

Umair Bari
On 5/9/06, Emmo ather [EMAIL PROTECTED] wrote:
Hello,In older version of freebpx if you write somethng manually in theconfiguration files it was flushed by amp, 
i.e. you can configure it throughthe interface only. Is this this thing still present in freepbx?_Don't just search. Find. Check out the new MSN Search!
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Re: [Asterisk-Users] CentOS 4.x and ooh323

2006-05-12 Thread Patrick

On Fri, 2006-05-12 at 19:50 +1200, Richard Scobie wrote:
[snip] 
  rpath /usr/local/lib -L./ooh323c/src -version-info 1:1:0  -lpthread
  make: rpath: Command not found
  make: [libchan_h323.la] Error 127 (ignored)
 
 My previous mail mentioned that this had been posted recently on the 
 list, however I was confusing it with the ooh323 list on Sourceforge.
 
 Unfortunately Sourceforge seems to be in the middle of a meltdown with 
 CVS access to projects broken and for over a week, no list mail is being 
 archived there, and I have deleted the mails I received relevent to this.
 
 So you may want to try on that list. This was the first post about it 
 before archiving stopped.
 
 http://sourceforge.net/mailarchive/forum.php?thread_id=10291962forum_id=43045

Thanks. I found that posting yesterday but it did not have an answer.
Other archived postings did not either.

Regards,
Patrick

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[Asterisk-Users] extension.conf for overlap

2006-05-12 Thread Nicolas LEGROS









Hi!!



Id like to confgure my extensions.conf
file in order to handle overlap!!

Is it possible?

What should be changed?



Thanks by advance

Nicolas L.






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Re: [Asterisk-Users] Please Help Me...Urgent

2006-05-12 Thread Umair Bari

Hello,

IMHO, there are 2 ways to do this,

1) You can connect your VoIP modem to your asterisk box using x100p FXO card, you'll need to get one and install it properly.

2) Get SIP/IAX account from any VoIP provider and use it with asterisk.

Hope this helps.

Regards,

Umair Bari
On 5/12/06, Crazy Boy [EMAIL PROTECTED] wrote:

Hi Friends,Thank you for your quick response. I have successfully implemented Intercom (Dialling within my office LAN) using Asterisk. To implement this, I am using X-Lite Softphone.
Now, I want to make calls to US using VoIP Asterisk. I think that there is no need of any external hardware to implement pure VoIP solution. Am I right?
I have registered with Vebtel (VoIP IP Telephony Service provider). They had given me one VoIP Modem called Voice Finder AP 200 and the below values:
Inbound Number: 123456789Public IP Number:
55.23.789.145Password: xyz(These values are dummy values)
Currently we are making US calls using VoIP provided by Vebtel. Now, I want to make US calls using this Vebtel service from Asterisk. How can I do this?
I am unable to understand where to give above mentioned values? What configuration files I should use to implement this using the Vebtel SIP provider? Do I need to provide any more values to implement this using Asterisk from Vebtel?
Waiting for your quick response. Thank you.
Regards,Chandra.

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[Asterisk-Users] RE: [PROBLEM] Still exist -- DTMF Tones, occures in Asterisk - Channelwide

2006-05-12 Thread Stefan Agethen


I don't see anything obviously wrong with your configs.



You don't want relaxdtmf.  That can cause the problem, not fix it.


Hi Eric,

at the begining - Thanks for your help.

relaxdtmf is not written in my config, so it should be at the default, i 
guess i remember default is yes ?
However, the dtmfmode should be the same, i think so, too, but my SNOMs 
working pretty well under RFC2833 , but my cheap Allnet´s

cannot handle dtmf unless it is in INFO Mode :(

But, i think, that cannot the problem. I have searched the Internet for 
talkoff but can only find the relaxdtmf option to increase it.


I cant understand why it happens on ALL kind of channels, if it would 
only happen on the SNOMs or something else, i could try INFO or 
something like this.


This problem exists - i guess few weeks before or after settling up from 
* 1.2.0 to * 1.2.5...


The configs are under construction, till now :-) Its Asterisk ;)

Any Idea ?
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Re: [Asterisk-Users] Asterisk TAPI - Outlook click2dial

2006-05-12 Thread Steve Davies


I don't know which version you downloaded, but if you can get the
source from CVS on Sourceforge, and build it yourself, you may have
more luck - The CVS version has code contributed from several sources,
and is slightly better that the packaged version.

Cheers,
Steve

On 5/12/06, Tomislav Vojvodic [EMAIL PROTECTED] wrote:

Oh.. :/ too bad..

I'll have to look at the source..

bye,


Tomislav


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of T. Shaw
Sent: Thursday, May 11, 2006 11:20 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial

 Yes, I have the exact same problem.
:(



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RE: [Asterisk-Users] Echo cancel: chan_misdn vs bristuff? HFC card vsexpensive card?

2006-05-12 Thread Chris Bagnall

 I've got a HFC ISDN card that I'm using with chan_misdn and 
 it basically behaves like crap. Echo is waaay worst then echo 
 I get TDM400 card, sound is choppy (there other side is 
 allays complaining about sound
 interruptions) and to top it all it detects fake DTMF's all the time.
 Is this a chan_misdn problem or is it a card problem? 

We have a number of sites running from 1-3 HFC-based cards in a machine, and
none of them have any significant echo at all. All ours are running with
zaphfc (part of the bristuff package). Might be worth giving that a try.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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Re: [Asterisk-Users] Bristuffed Asterisk: Hangup problems

2006-05-12 Thread stoffell


On 5/11/06, Tim Robinson [EMAIL PROTECTED] wrote:

There is a lot of junk in your zapata.conf that you do not need, as it
relates to analogue lines.  This might be causing confusion?


I have tried a similary config to yours, doesn't helps. I haven't got
this problem on an E1, just on the newer bristuff'd packages.

I have sent an email to junghanns.net about this, but haven't received
an answer yet. If I do receive anything, I'll post it back to the
thread.

If there are any other things I might check, please let me know..

cheers
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Re: [Asterisk-Users] Please Help Me...Urgent

2006-05-12 Thread Alejandro Vargas


2006/5/12, Crazy Boy [EMAIL PROTECTED]:

 I am unable to understand where to give above mentioned values? What
configuration files I should use to implement this using the Vebtel SIP
provider? Do I need to provide any more values to implement this using
Asterisk from Vebtel?


In addition to your username/phone number/password given to you by
your provider, you need to know what protocol and codecs is the voip
adapter using. Let's supose it is using sip. Then you configure your
asterisk to access a sip trunk like is described in many pages.

Checking google I see this:

2. Services   Products offered by   Vebtel?

Vebtel offers wide solutions to cut down your ISD telephone bills.
Vebtel can operate and interoperate with multiple VoIP products like
Soft phone, Gateways, IP Phone and Calling Cards operating on SIP and
H.323. For more Product and Service information click here ...

Some voip providers uses mgcp or even the old h323. This should work
but will be more difficult to configure because it is uncommon.

--
Alejandro Vargas
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Re: [Asterisk-Users] Echo cancel: chan_misdn vs bristuff? HFC card vs expensive card?

2006-05-12 Thread Woodoo People .pGa!

 I've got a HFC ISDN card that I'm using with chan_misdn and it basically 
 behaves like crap. Echo is waaay worst then echo I get TDM400 card, 
 sound is choppy (there other side is allays complaining about sound 
 interruptions) and to top it all it detects fake DTMF's all the time.
 
 Is this a chan_misdn problem or is it a card problem? I really need to 
 get this fix and I need to know the way to go. I don't want to throw 
 money at a better card if the card is not the issue but if that's the 
 only solution, I'll need to order the card ASAP!

i'm using 1port (billion bipac), quad and octoBRI cards from beronet.
all of them working nice, beronet recommend to use kernel 2.6.12+ and
asterisk 1.2.x and also newest misdn-mqueue from www.beronet.com
-- 
WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com
[EMAIL PROTECTED]@RedHat.users
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Re: [Asterisk-Users] Echo cancel: chan_misdn vs bristuff? HFC card vs expensive card?

2006-05-12 Thread Cosmin Prund


Woodoo People .pGa! wrote:
I've got a HFC ISDN card that I'm using with chan_misdn and it basically 
behaves like crap. Echo is waaay worst then echo I get TDM400 card, 
sound is choppy (there other side is allays complaining about sound 
interruptions) and to top it all it detects fake DTMF's all the time.


Is this a chan_misdn problem or is it a card problem? I really need to 
get this fix and I need to know the way to go. I don't want to throw 
money at a better card if the card is not the issue but if that's the 
only solution, I'll need to order the card ASAP!



i'm using 1port (billion bipac), quad and octoBRI cards from beronet.
all of them working nice, beronet recommend to use kernel 2.6.12+ and
asterisk 1.2.x and also newest misdn-mqueue from www.beronet.com
  
Seeing the names on this list I realize I've tried lots and lots of 
different things. I'm running kernel 2.6.15.11 so I'm above 12. 
Unfortunately the latest misdn-mqueue does not compile on my system, it 
issues all sorts of blah-blah that I'm interpreting in only one way: 
there's a problem with the parameters the Makefile passes to the 
compiler (the .h files where the error manifests itself are part of 
Asterisk and compile fine when compiled with Asterisk itself).


Those are the errors I get:

./create_config.sh /usr/include
Checking Asterisk version...
* found 'struct ast_channel_tech'
* found 'ast_bridged_channel'
* found 'ast_bridge_result'
* found bridge with timeoutms
* ast_dsp_process() without 'needlock'
* found 'struct ast_callerid'
* found 'struct timeval delivery'
* found 'transfercapability'
* found 'ast_config_load'
* found 'AST_CONTROL_HOLD'
* found 'devicestate.h'
* found 'strings.h'
* no 'type' in ast_channel
* found stringfield in ast_channel
config.h complete.
gcc -pipe -fPIC -Wall -Wmissing-prototypes -Wmissing-declarations -g  
-I/usr/include -D_REENTRANT -D_GNU_SOURCE  -O6 -march=i686  
-Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO   -c -o 
chan_capi.o chan_capi.c

In file included from /usr/include/asterisk/utils.h:36,
   from /usr/include/asterisk/cdr.h:48,
   from /usr/include/asterisk/channel.h:113,
   from chan_capi.c:23:
/usr/include/asterisk/strings.h:264: error: syntax error before 
__extension__

/usr/include/asterisk/strings.h:264: error: syntax error before ';' token
/usr/include/asterisk/strings.h:264: error: `__len' undeclared here (not 
in a function)
/usr/include/asterisk/strings.h:264: error: initializer element is not 
constant

/usr/include/asterisk/strings.h:264: error: syntax error before if
/usr/include/asterisk/strings.h:264: error: redefinition of '__retval'
/usr/include/asterisk/strings.h:264: error: previous definition of 
'__retval' was here

/usr/include/asterisk/strings.h:264: error: syntax error before const
/usr/include/asterisk/strings.h:264: error: syntax error before '}' token
/usr/include/asterisk/strings.h:280: error: conflicting types for 'strtoq'



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[Asterisk-Users] issue has arisen

2006-05-12 Thread scott

Hi aLL

I have an [EMAIL PROTECTED] box running.
When i register via SIP to the box with 2 phones both behind the same firewall 
the registration goes through fine and I can see in the realtime database AND 
that an IP and port has been entered for each extension, its obviously same IP 
(the firewall) but different ports for each registration.

When i try to make a call from one of the extensions it fails but the other one 
is fine.
If i unregister the working extension and remove the IP and port from the 
realtime table the other phone works fine.

Does anyone know how I can overcome this?

I have another normal asterisk server with 3 phones behind the same firewall 
but entrys are not realtime they are direct in the sip.conf and i dont have 
this problem.

Many Thanks
Scott
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Re: [Asterisk-Users] Echo cancel: chan_misdn vs bristuff? HFC card vsexpensive card?

2006-05-12 Thread Cosmin Prund


Chris Bagnall wrote:
I've got a HFC ISDN card that I'm using with chan_misdn and 
it basically behaves like crap. Echo is waaay worst then echo 
I get TDM400 card, sound is choppy (there other side is 
allays complaining about sound

interruptions) and to top it all it detects fake DTMF's all the time.
Is this a chan_misdn problem or is it a card problem? 



We have a number of sites running from 1-3 HFC-based cards in a machine, and
none of them have any significant echo at all. All ours are running with
zaphfc (part of the bristuff package). Might be worth giving that a try.

Regards,

Chris
  
Thanks for the info, I'm compiling bristuff 0.3 right now, hope it 
works. I'll need to wait for the compile to finish to see how it works 
with my kernel (as my kernel has been patched for mISDN and I do not 
know how that plays with bristuff).

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Re: [Asterisk-Users] mISDN trouble with a HFC Cologne card, Asterisk Asterisk 1.2.4 on Linux 2.6.16.11 - incoming DTMF detection

2006-05-12 Thread Michel Koenen


I also hadan issueto getmISDNworking with HFC cards without problems.
Therefor I switched to zaphfc (use bristuff), this is working perfectly with HFC cards. It does everything I need including MSN support and without problems, even with multiple HFC cards.

So my advice is to get rid of mISDN and to switch to zaphfc

Regards
Michel
From: Cosmin Prund 
[EMAIL PROTECTED]Subject: [Asterisk-Users] mISDN trouble with a HFC Cologne card,   Asterisk Asterisk 1.2.4 on Linux 
2.6.16.11 - incoming DTMF detectionHello everyone. I've got this really annoying HFC Cologne card (orhowever I should call it - a single channel ISDN card based on the HFCchipset).It wrongfully detects lots and lots and lots of incoming DTMFs, to the
point the card is not usable.Here's a sample out of CLI:P[ 1] I IND :DTMF_TONE oad:206361 dad:520101P[ 1] -- mode:TE cause:16 ocause:16 rad: cad:P[ 1] -- facility:FAC_NONE out_facility:FAC_NONE
P[ 1] -- info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0P[ 1] -- screen:0 -- pres:0P[ 1] -- channel:1 caps:Speech pi:2 keypad:P[ 1] -- urate:0 rate:16 mode:0 user1:0P[ 1] -- pid:1 addr:50010102 l3id:30001
P[ 1] -- b_stid:10010100 layer_id:50010180P[ 1] -- bc_state:BCHAN_ACTIVATEDP[ 1] -- DTMF:*What's this all about? Is there anything I can do about it?
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[Asterisk-Users] SCCP audio problems

2006-05-12 Thread Juanjo Portela


Dear Colleagues,

I have 2 phones Cisco 12SP+ connected to my asterisk box 1.2.6 and
SCCP channel version: 20060408.

When a call is generated no audio pass through the phones, neither if
i call from a 12SP+ to another nor calling between 12SP and other
phone (ex. an x-lite). Only sometimes works fine.

What am I doing wrong?

Thank you for your help.
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RE: [Asterisk-Users] Dialling a DUNDi Route

2006-05-12 Thread Watkins, Bradley

I'm not sure if you have considered this, but if you were using SIP
between the Asterisk servers you can definitely achieve this using
X-headers.

Regards,
- Brad

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Thursday, May 11, 2006 11:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Dialling a DUNDi Route

Patrick,
 
Dug all day... found nothing!

-Original Message- 
From: Patrick [mailto:[EMAIL PROTECTED] 
Sent: Thu 5/11/2006 3:11 PM 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Cc: 
Subject: RE: [Asterisk-Users] Dialling a DUNDi Route



On Thu, 2006-05-11 at 10:33 -0600, Douglas Garstang wrote:
[snip]
 When you IAX trunk a call from Asterisk A to Asterisk B, you
can't pass the ring time and ring options of the original SIP call
between servers.

Iirc you can pass variables on the IAX link to the other side.
Maybe you
can use those settings to define the ringtime etc. Don't recall
how to
pass them though so you need to do some digging there.

Regards,
Patrick

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contains information that may be confidential. Unless you are the named 
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Re: [Asterisk-Users] ATXFER

2006-05-12 Thread Josué Conti

Eric, thank you very much. But It could help in this case me?

Regards

Josué
2006/5/12, Eric ManxPower Wieling [EMAIL PROTECTED]:
Josué Conti wrote: Dinesh, very obliged for the attention. I am using version 1.0.9 of asterisk
 and it is really all good with this version, only this case of atxfer that it does not function. The function DYNAMIC_FEATURE = to atxfer in my [ globals ] of extensions.conf functions in version 
1.0.9? It could help in this case me? Best Regards1.0.x does not support this.--Now accepting new clients in Birmingham, Atlanta, Huntsville,Chattanooga, and Montgomery.___
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[Asterisk-Users] email - fax gateway with billing possibilities?

2006-05-12 Thread Roy Sigurd Karlsbakk


hi

does anyone have an idea how it could be possible to do email - fax  
gatewaying with asterisk + app_txfax, but still keep track of who  
sent the fax? i've thought a little about smtp auth, but it doesn't  
look too easy to integrate smoothly with asterisk


roy

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[Asterisk-Users] Hint priority

2006-05-12 Thread richard Coco


Hi all,

i am desperating, trying to configure an OptiPoint410
with the hint priority.

Here what i have...

OptiPoint410std- exten 2001
X-Lite - exten 2002

But unfortunately no LED ON on my OptiPoint410

sip.conf
[2001]
type=friend
context=local
host=dynamic
dtmfmode=rfc2833
incominglimit=1
notifyringing=yes
subscribecontext=default
disallow=all
allow=alaw
allow=ulaw

[2002]
type=friend
context=local
host=dynamic
dtmfmode=rfc2833
incominglimit=1
notifyringing=yes
subscribecontext=default
disallow=all
allow=alaw
allow=ulaw

extensions.conf
[default]
exten = 2001,hint,SIP/2001
exten = 2002,hint,SIP/2002

[local]
exten = 2001,1,Dial(SIP/2001,10,tr)
exten = 2001,2,HangUp

exten = 2002,1,Dial(SIP/2002,10,tr)
exten = 2002,2,HangUp

Has anyone managed get hint working with an
OptiPoint4x0.

thx in advance


__
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Re: [Asterisk-Users] monitoring sangoma cards via snmp

2006-05-12 Thread Sean Cook


[EMAIL PROTECTED] wrote:

Hello,

Digium does not provide snmp support to monitor their
cards !

  
That's like saying Toyota doesn't provide gas with their cars.  You can 
setup snmp with in linux and have it execute commands that you want to 
determine whether or not the hardware is functioning as you wish.  
Hardware is hardware...  Intel doesn't provide snmp for my motherboard 
either...



Anybody has tried Sangoma product A104 Quad T1/E1 or
others ?
  
I don't think you are going to find exactly what you are looking for 
with out purchasing an appliance...


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[Asterisk-Users] Sangoma A200D problem

2006-05-12 Thread Dr. Michael J. Chudobiak


Hi all,

I've been having problems with my A20002D lately - callers from the PSTN 
don't hear me when I answer, but I hear them. Disabling echo 
cancellation in zapata.conf brings the audio (and echo) back. This used 
to work fine, until two days ago.


The only weird thing in the logs is this:

May 12 07:42:53 steerpike wan_ecd: wp1ec: The H100 slave has lost its 
framing on the bus!
May 12 07:42:53 steerpike wan_ecd: wp1ec: The CT_C8_A clock behavior 
does not conform to the H.100 spec!


Is this a problem? (The server is an HP ProLiant DL140 G2).

Has anyone else seen this type of erratic problem?

I've tried re-seating the A20002D card, the FXO plug-ins, and the 
echo-canceller plug-ins. Right now it works ... we'll see if it stays 
that way.


- Mike

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[Asterisk-Users] S100-FX v2 audio quality

2006-05-12 Thread Ben Holt


Hello,

In a fit of optimism I recently purchased a X100-FX v2 
(http://x100p.com/products_2.htm) despite the lack of reviews I was able 
to find on the device.  The feature set made it hard to resist.  I have 
since been experiencing audio quality issues with it.


Do any other mailing list members have experience with this ATA?  If so, 
could you let me know if you are satisfied with its audio quality.  At 
this point I don't know if I have a bum unit, a configuration problem, 
or am having a typical experience.


Many thanks,

- Ben
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[Asterisk-Users] Speex fans?

2006-05-12 Thread Dr. Michael J. Chudobiak


Hi all,

I've been testing various codecs to eliminate choppiness that I 
sometimes get on my Asterisk IAX2  DSL  provider (Exgn) connections, 
and Speex seems to work the best, so far - but Speex seems oddly unpopular.


Can anyone share their experiences with Speex (good and bad)? Is anyone 
using it in a production environment?


I like the variable bit rate and packet loss concealment features...


- Mike
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[Asterisk-Users] voice mail notification

2006-05-12 Thread Ever Zalazar




Hello, there is a way to send notification(not 
email) when it's received an voice mail? Maybe a SIP message to 
inform?


Best REgards


Ever Zalazar
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RE: [Asterisk-Users] features.conf *1 Call Recording




Yes. I did.

David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodatasolutions.com

Tel: (519) 963-3020
Fax: (519) 451-6615

 Lead, follow or get out of the 
way! 


This message has originated from Autodata Solutions. The attached material is 
the Confidential and Proprietary Information of Autodata Solutions. This email 
and any files transmitted with it are confidential and intended solely for the 
use of the individual or entity to whom they are addressed. If you have received 
this email in error please delete this message and notify the Autodata system 
administrator at [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED]



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Giridhar 
Reddy BandiSent: Friday, May 12, 2006 3:41 AMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] features.conf *1 Call Recording
did you include automon = *1 in your features.conf ?? it 
should be somthing like this [featuremap]automon = 
*1 
--Giridhar Bandi 
On 5/12/06, Dave 
Morrow [EMAIL PROTECTED] 
wrote:
Thanks 
  for the response.How would I change the DTMF transfer 
  mode?David MorrowTechnical Systems LeadAutodata Solutions 
  Company[EMAIL PROTECTED] http://www.autodatasolutions.comTel: 
  (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of 
  the way! This message has originated from Autodata Solutions. The 
  attached material is the Confidential and Proprietary Information of 
  AutodataSolutions. This email and any files transmitted with it are 
  confidentialand intended solely for the use of the individual or entity to 
  whom they are addressed. If you have received this email in error please 
  deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] 
  mailto: [EMAIL PROTECTED]-Original 
  Message-From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]] 
  On Behalf Of FabioSent: Thursday, May 11, 2006 6:55 PMTo: Asterisk 
  Users Mailing List - Non-Commercial Discussion Subject: RE: 
  [Asterisk-Users] features.conf *1 Call Recordingif you ar using SIP 
  clients, try changing DTMF transfer mode.For test use sip 
  debugon your * console, then place a call and watch the output. In INFO or 
  rfc2833 mode you must see the codes like SIP messages. If you are 
  usinginband transfer mode (DTMF codes aretransferred like 
  sounds) you don'tsee the codes.Also, try adjusting 
  featuredigittimeout in features.conf :[general]featuredigittimeout 
  = 2000 ; 2 secondsbecause the default 500ms is a very short 
  time.Fabio Olaechea3Tech SRLCalle 48 Nro 632, Of. 
  67.La Plata, CP B1900AMZBuenos Aires, Argentina. Tel. +54 221 445 
  0244 Ext. 301Fax. +54 221 445 0245www.trestech.com.ar-Mensaje 
  original-De: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]]En 
  nombre de Dave MorrowEnviado el: Miercoles, 10 de Mayo de 2006 02:48 
  p.m.Para: Asterisk Users Mailing List - Non-Commercial 
  DiscussionAsunto: RE: [Asterisk-Users] features.conf *1 Call 
  RecordingOK. You lost me.David MorrowTechnical 
  Systems LeadAutodata Solutions Company [EMAIL PROTECTED]http://www.autodatasolutions.comTel: 
  (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of 
  the way!  This message has originated from Autodata Solutions. The 
  attachedmaterial is the Confidential and Proprietary Information of 
  AutodataSolutions. This email and any files transmitted with it are 
  confidential and intended solely for the use of the individual or entity 
  to whom theyare addressed. If you have received this email in error please 
  deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] 
  mailto:[EMAIL PROTECTED]-Original 
  Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] 
  On Behalf Of AlejandroVargasSent: Wednesday, May 10, 2006 10:29 AM 
  To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: 
  Re: [Asterisk-Users] features.conf *1 Call Recording2006/5/10, Dave 
  Morrow [EMAIL PROTECTED] 
  : I am attempting to setup Asterisk to allow me to press *1 
  while in a call to use automon to record the call but have had 
  absolutely no success.Is there a trick to this?May 
  be a problem with the way you are sending the dialtones. Try sending as 
  data.--Alejandro 
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Re: [Asterisk-Users] S100-FX v2 audio quality

2006-05-12 Thread Bill Peck

On 5/12/06, Ben Holt [EMAIL PROTECTED] wrote:
Hello,In a fit of optimism I recently purchased a X100-FX v2(http://x100p.com/products_2.htm) despite the lack of reviews I was ableto find on the device.The feature set made it hard to resist.I have
since been experiencing audio quality issues with it.Do any other mailing list members have experience with this ATA?If so,could you let me know if you are satisfied with its audio quality.Atthis point I don't know if I have a bum unit, a configuration problem,
or am having a typical experience.Many thanks,I can't even get mine to work. :-( Can you share your configuration for it?


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RE: [Asterisk-Users] features.conf *1 Call Recording




All I see when I press *1 is 

 -- Attempting native bridge of 
SIP/8001-252e and SIP/3020-5171
I still cannot make this work.

David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodatasolutions.com

Tel: (519) 963-3020
Fax: (519) 451-6615

 Lead, follow or get out of the 
way! 


This message has originated from Autodata Solutions. The attached material is 
the Confidential and Proprietary Information of Autodata Solutions. This email 
and any files transmitted with it are confidential and intended solely for the 
use of the individual or entity to whom they are addressed. If you have received 
this email in error please delete this message and notify the Autodata system 
administrator at [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED]



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Dave 
MorrowSent: Friday, May 12, 2006 8:39 AMTo: Asterisk Users 
Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] 
features.conf *1 Call Recording

Yes. I did.

David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodatasolutions.com

Tel: (519) 963-3020
Fax: (519) 451-6615

 Lead, follow or get out of the 
way! 


This message has originated from Autodata Solutions. The attached material is 
the Confidential and Proprietary Information of Autodata Solutions. This email 
and any files transmitted with it are confidential and intended solely for the 
use of the individual or entity to whom they are addressed. If you have received 
this email in error please delete this message and notify the Autodata system 
administrator at [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED]



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Giridhar 
Reddy BandiSent: Friday, May 12, 2006 3:41 AMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] features.conf *1 Call Recording
did you include automon = *1 in your features.conf ?? it 
should be somthing like this [featuremap]automon = 
*1 
--Giridhar Bandi 
On 5/12/06, Dave 
Morrow [EMAIL PROTECTED] 
wrote: 
Thanks 
  for the response.How would I change the DTMF transfer 
  mode?David MorrowTechnical Systems LeadAutodata Solutions 
  Company[EMAIL PROTECTED] http://www.autodatasolutions.comTel: 
  (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of 
  the way! This message has originated from Autodata Solutions. The 
  attached material is the Confidential and Proprietary Information of 
  AutodataSolutions. This email and any files transmitted with it are 
  confidentialand intended solely for the use of the individual or entity to 
  whom they are addressed. If you have received this email in error please 
  deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] 
  mailto: [EMAIL PROTECTED]-Original 
  Message-From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]] 
  On Behalf Of FabioSent: Thursday, May 11, 2006 6:55 PMTo: Asterisk 
  Users Mailing List - Non-Commercial Discussion Subject: RE: 
  [Asterisk-Users] features.conf *1 Call Recordingif you ar using SIP 
  clients, try changing DTMF transfer mode.For test use sip 
  debugon your * console, then place a call and watch the output. In INFO or 
  rfc2833 mode you must see the codes like SIP messages. If you are 
  usinginband transfer mode (DTMF codes aretransferred like 
  sounds) you don'tsee the codes.Also, try adjusting 
  featuredigittimeout in features.conf :[general]featuredigittimeout 
  = 2000 ; 2 secondsbecause the default 500ms is a very short 
  time.Fabio Olaechea3Tech SRLCalle 48 Nro 632, Of. 
  67.La Plata, CP B1900AMZBuenos Aires, Argentina. Tel. +54 221 445 
  0244 Ext. 301Fax. +54 221 445 0245www.trestech.com.ar-Mensaje 
  original-De: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]]En 
  nombre de Dave MorrowEnviado el: Miercoles, 10 de Mayo de 2006 02:48 
  p.m.Para: Asterisk Users Mailing List - Non-Commercial 
  DiscussionAsunto: RE: [Asterisk-Users] features.conf *1 Call 
  RecordingOK. You lost me.David MorrowTechnical 
  Systems LeadAutodata Solutions Company [EMAIL PROTECTED]http://www.autodatasolutions.comTel: 
  (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of 
  the way!  This message has originated from Autodata Solutions. The 
  attachedmaterial is the Confidential and Proprietary Information of 
  AutodataSolutions. This email and any files transmitted with it are 
  confidential and intended solely for the use of the individual or entity 
  to whom theyare addressed. If you have received this email in error please 
  deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] 
  mailto:[EMAIL PROTECTED]-Original 
  Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] 
  On Behalf Of AlejandroVargasSent: Wednesday, May 10, 2006 10:29 AM 
  To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: 
  Re: [Asterisk-Users]

RE: [Asterisk-Users] MeetME Conferencing

2006-05-12 Thread Damon Estep

Josh,

Thank you!

I think the AGI could be bypassed by doing a realtime() to get the PIN from 
mySQL, also returning the variable that defines admin or user and jumping in 
the dialplan accordingly. Otherwise I would just end up having the AGI do the 
query because there is a need to store the users in the database to facilitate 
easy management.

The admin menu and marked user options seem to be the key to making this work, 
so I will play with those.





-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josh McAllister
Sent: Friday, May 12, 2006 2:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] MeetME Conferencing

Ok, the script below (meetme.agi) will prompt for a valid pin up to 3 times. If 
the pin matches one of the defined Admin pins, it will set the dialplan 
priority to 10 and exit, if User, sets to 20 and exits. Otherwise Hangs up.

In the case of admin, these MeetMe options are used:
a - Admin mode
A - Marked mode
c - Announce number of participants (optional of course)
s - Present Admin menu by pressing '*'
x - close conf when last marked user leaves.

In the case of user:
c s x are used as above, but we add:
w - wait until marked user enters. (Plays MoH until then)

The dialplan assumes you have a static pinless conference setup as conf #10.

extensions.conf:
exten = 5552323,1,Wait(1)
exten = 5552323,2,Answer()
exten = 5552323,3,AGI(meetme.agi)
exten = 5552323,4,NoOp(Invalid Pin)
exten = 5552323,5,Hangup()

exten = 5552323,10,NoOp(Admin Pin)
exten = 5552323,11,MeetMe(10,aAcsx)
exten = 5552323,12,Hangup()

exten = 5552323,20,NoOp(User Pin)
exten = 5552323,21,MeetMe(10,cswx)
exten = 5552323,22,Hangup()



The script of course requires the Asterisk::AGI module.

meetme.agi:

#!/usr/bin/perl
use Asterisk::AGI;
my $AGI = new Asterisk::AGI;
my $input = { %{$AGI-ReadParse()} };

#our $DEBUG = 1; 

my @UserPins = ('1','2');
my @AdminPins = ('9','8');

my $mode = collectPin($AGI,5);

$AGI-verbose(collectPin got '$mode') if $DEBUG;

if ($mode eq 'Admin') {
   $AGI-set_priority(10);
} elsif ($mode eq 'User') {
   $AGI-set_priority(20);
} else {
   $AGI-stream_file(goodbye,'');
   $AGI-hangup;
}

exit;

sub collectPin {
   my $AGI = shift;
   my $maxdigits = shift;

   my $tries = 0;

   #Three tries to select an existing pin.
   while ($tries  3) {
  $AGI-stream_file(please-try-again,'') if $tries  0;
  $tries++;
  my $pin = $AGI-get_data('enter-conf-pin-number', 1, $maxdigits);
  $AGI-verbose(Got PIN $pin.) if $DEBUG;
  next unless $pin  0;

  if ( grep(/^$pin$/, @AdminPins) ) {
 $AGI-stream_file(pin-number-accepted,'');
 return 'Admin';
  } elsif ( grep(/^$pin$/, @UserPins) ) {
 $AGI-stream_file(pin-number-accepted,'');
 return 'User';
  } else {
 $AGI-stream_file(conf-invalidpin,'');
  }
   }

   return undef;
}


What can I say, I was bored.

Enjoy,

Josh McAllister

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Thursday, May 11, 2006 10:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] MeetME Conferencing

Static configs for the conference rooms are not an issue.

The main goal is to allow the moderator to determine when the conference 
“starts” by having all participants hearing MOH until the moderator starts the 
interactive call with a  PIN known only to the moderator, and then allowing the 
moderator (and only the moderator) to kick out all users from the keypad when 
the call is over.

An additional benefit would be gained if authenticate() or realtime() app 
commands could be used against a mysql database for the participant and 
moderator pins so an app could be written easily to allow changing of the PINS 
in the database.


From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Austin
Sent: Thursday, May 11, 2006 10:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] MeetME Conferencing

I believe you can accomplish this with a well crafted dialplan.
 
If you did not have the restriction against out of tree modules, I would
recommend an app that strores the conference details in a database
and would allow just this kind of control.
 
Dan


From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Thursday, May 11, 2006 4:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] MeetME Conferencing
Not opposed to paying someone that can do it right ☺
 
As far as “coding” goes, you mean create the dialplan entries, not modify the 
meetme source, correct?
 
Our application requires that this can be done in 1.2 release, not trunk and 
not with an add-in that is not part of 1.2
 
If you have done it

[Asterisk-Users] call parked / MOH

Hello,

How can I park a call or put on hold a caller from an
analogue to sip agents ?

  PSTN===FXO/asterisk=sip agents

When I press hold key or #800 the channel is hangup ??

Harry
Regards 








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[Asterisk-Users] Re: Sangoma A200D problem

2006-05-12 Thread Andre Courchesne - Consultant


Hi,

 Last time I had this problem was following a unclean powerdown and the 
solution was:

   - Kill Asterisk
   - Stop wanpipe
   - cd /etc/wanpipe/wan_ec
   - In there there should be 2 files:
wan_ec_pid
wan_ec_socket=
   - Delete those files
   - Perform a reboot of your system

--
Andre Courchesne

[EMAIL PROTECTED] wrote:

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To subscribe or unsubscribe via the World Wide Web, visit
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When replying, please edit your Subject line so it is more specific
than Re: Contents of Asterisk-Users digest...


Today's Topics:

   1. Re: Echo cancel: chan_misdn vs bristuff? HFC card vs
  expensive card? (Woodoo People .pGa!)
   2. Re: Echo cancel: chan_misdn vs bristuff? HFC card vs
  expensive card? (Cosmin Prund)
   3. issue has arisen (scott)
   4. Re: Echo cancel: chan_misdn vs bristuff? HFC card vsexpensive
  card? (Cosmin Prund)
   5. Re: mISDN trouble with a HFC Cologne card,Asterisk Asterisk
  1.2.4 on Linux 2.6.16.11 - incoming DTMF detection (Michel Koenen)
   6. SCCP audio problems (Juanjo Portela)
   7. RE: Dialling a DUNDi Route (Watkins, Bradley)
   8. Re: ATXFER ( Josu? Conti )
   9. email - fax gateway with billing possibilities?
  (Roy Sigurd Karlsbakk)
  10. Hint priority  (richard Coco)
  11. Re: monitoring sangoma cards via snmp (Sean Cook)
  12. Sangoma A200D problem (Dr. Michael J. Chudobiak)
  13. S100-FX v2 audio quality (Ben Holt)
  14. Speex fans? (Dr. Michael J. Chudobiak)
  15. voice mail notification (Ever Zalazar)
  16. RE: features.conf *1 Call Recording (Dave Morrow)
  17. Re: S100-FX v2 audio quality (Bill Peck)
  18. RE: features.conf *1 Call Recording (Dave Morrow)


--

Message: 1
Date: Fri, 12 May 2006 12:51:32 +0200
From: Woodoo People .pGa! [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Echo cancel: chan_misdn vs bristuff? HFC
card vs expensive card?
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii

  
I've got a HFC ISDN card that I'm using with chan_misdn and it basically 
behaves like crap. Echo is waaay worst then echo I get TDM400 card, 
sound is choppy (there other side is allays complaining about sound 
interruptions) and to top it all it detects fake DTMF's all the time.


Is this a chan_misdn problem or is it a card problem? I really need to 
get this fix and I need to know the way to go. I don't want to throw 
money at a better card if the card is not the issue but if that's the 
only solution, I'll need to order the card ASAP!



i'm using 1port (billion bipac), quad and octoBRI cards from beronet.
all of them working nice, beronet recommend to use kernel 2.6.12+ and
asterisk 1.2.x and also newest misdn-mqueue from www.beronet.com
  


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Re: [Asterisk-Users] S100-FX v2 audio quality

2006-05-12 Thread Gareth Blades

I just bought a couple of these units. It seems to work fine but I could
not really test it as the phones were too close together so could not
get a clear idea of the call quality.

Phoning comedian mail seemed fine and certenly acceptible considering
the gsm codec was being used.

One minor annoyance is that it is configured with an absolute setting
for the registry interval and does not pay any attention to what
asterisk says. You therefore need to make sure it does not exceed the
asterisk setting otherwise you get continuous asterisk warnings.

Its missing a couple of features though :-

1) You can only configure the callerid number and not the name.

2) Message waiting is not supported.

On Fri, 2006-05-12 at 13:40, Bill Peck wrote:
 
 On 5/12/06, Ben Holt [EMAIL PROTECTED] wrote:
 Hello,
 
 In a fit of optimism I recently purchased a X100-FX v2
 (http://x100p.com/products_2.htm) despite the lack of reviews
 I was able
 to find on the device.  The feature set made it hard to
 resist.  I have 
 since been experiencing audio quality issues with it.
 
 Do any other mailing list members have experience with this
 ATA?  If so,
 could you let me know if you are satisfied with its audio
 quality.  At
 this point I don't know if I have a bum unit, a configuration
 problem, 
 or am having a typical experience.
 
 Many thanks,
 
 I can't even get mine to work. :-(Can you share your configuration
 for it?
 
 
 
 
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RE: [Asterisk-Users] Speex fans?

2006-05-12 Thread Andrew Kirch

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Dr. Michael J. Chudobiak
 Sent: Friday, May 12, 2006 8:20 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Speex fans?

 Hi all,

 I've been testing various codecs to eliminate choppiness that I
 sometimes get on my Asterisk IAX2  DSL  provider (Exgn)
connections,
 and Speex seems to work the best, so far - but Speex seems oddly
 unpopular.

 Can anyone share their experiences with Speex (good and bad)? Is
anyone
 using it in a production environment?

 I like the variable bit rate and packet loss concealment features...

 - Mike

I believe the tradeoff is that though it's compressed it uses a bit more
bandwidth and a bit more CPU.  Because of the redundancy to remove chop,
it has greater overhead on almost all counts.  I have no issue using it,
and frequently use it for asterisk  asterisk trunking where bandwidth
is insufficient for uLaw.

Andrew
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URGENT please [Asterisk-Users] call parked / MOH




 Hello,
 
 How can I park a call or put on hold a caller from
 an
 analogue to sip agents ?
 
   PSTN===FXO/asterisk=sip agents
 
 When I press hold key or #800 the channel is hangup
 ??
 
 Harry
 Regards 






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[Asterisk-Users] Automon Filenames

2006-05-12 Thread David Sampson









Can someone give me some direction on automon filenames? I
would like them to be the dialed number if possible. I saw a patch available
for changing this but havent quite figured out how to use it.


Can someone point me in the right direction?

Thanks,


Dave






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Re: [Asterisk-Users] S100-FX v2 audio quality


Same problem with audio quality.  Got rid of them.  Also the context
line only allowed 12 characters and we need more than that for some
installations, I didn't want to have to rename 100 contexts to less
than 12 characters.

On 5/12/06, Gareth Blades [EMAIL PROTECTED] wrote:

I just bought a couple of these units. It seems to work fine but I could
not really test it as the phones were too close together so could not
get a clear idea of the call quality.

Phoning comedian mail seemed fine and certenly acceptible considering
the gsm codec was being used.

One minor annoyance is that it is configured with an absolute setting
for the registry interval and does not pay any attention to what
asterisk says. You therefore need to make sure it does not exceed the
asterisk setting otherwise you get continuous asterisk warnings.

Its missing a couple of features though :-

1) You can only configure the callerid number and not the name.

2) Message waiting is not supported.

On Fri, 2006-05-12 at 13:40, Bill Peck wrote:

 On 5/12/06, Ben Holt [EMAIL PROTECTED] wrote:
 Hello,

 In a fit of optimism I recently purchased a X100-FX v2
 (http://x100p.com/products_2.htm) despite the lack of reviews
 I was able
 to find on the device.  The feature set made it hard to
 resist.  I have
 since been experiencing audio quality issues with it.

 Do any other mailing list members have experience with this
 ATA?  If so,
 could you let me know if you are satisfied with its audio
 quality.  At
 this point I don't know if I have a bum unit, a configuration
 problem,
 or am having a typical experience.

 Many thanks,

 I can't even get mine to work. :-(Can you share your configuration
 for it?




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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Re: [Asterisk-Users] MeetME Conferencing

2006-05-12 Thread Mike Clark


Damon Estep wrote:


Can anyone point me to a sample or information on using MeetMe like this?

 

Conference room is set up with 2 PINs, one for the moderator and one 
for the participants.


Participants get music until the moderator joins (to avoid wild, 
un-moderated tangents).


Call is ended and all participants are kicked out when the moderator 
leaves (or the moderator can kick everyone out via phone keypad).


 


Asking too much, or simple stuff?

 


Damon

 

Latest version of Web-MeetMe will do this, but it is definitely of the 
add-on variety.


You can do it pure dial plan if you are willing to have a menu that says 
Press 1 to join as admin, Press 2 to join as participant. Then you 
simply set the meetme options accordingly. The A (mark user) combined 
with a (admin) and w (wait for marked user) are the key options 
here. With a little agi magic, you could have a single entry point 
without the user specifying whether they were admin or participant.


Mike Clark


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RE: [Asterisk-Users] Paging and Auto Answer on Grandstream GXP2000

2006-05-12 Thread Forrest Beck

Asterisk 1.2.7.1 and Zaptel 1.2.5

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
Sent: Thursday, May 11, 2006 6:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Paging and Auto Answer on Grandstream GXP2000

What version of Asterisk?

On 5/11/06, Forrest Beck [EMAIL PROTECTED] wrote:

 I am looking to setup paging using the auto answer feature on the
 Grandstream GXP2000.  I am thinking I will follow the method as described
 here:

 http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page

 I will setup the 4th account on the phone to auto answer.

 Does anyone else have a method that works better?  I also looked at the
 allpage AGI written on Voip-Info.  But it seems to dial all extensions,
even
 the ones I don't want to use for Auto Answer.

 I really would like a way to group the phones instead of having them all
 listed in a dial command.

 exten =
 7999,2,Page(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL 
 PROTECTED]/nLocal/interal
 [EMAIL PROTECTED]|)

 Thanks!
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-- 
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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[Asterisk-Users] rxfax problem

2006-05-12 Thread Woodoo People .pGa!

Hi!

Anyone meet with the following problem?

May 12 15:51:44 WARNING[14399] channel.c: Unable to find a codec translation 
path from ulaw to unknown
May 12 15:51:44 WARNING[14399] app_txfax.c: Unable to restore read format on 
'SIP/neopost1-8083'
May 12 15:51:44 WARNING[14399] channel.c: Unable to find a codec translation 
path from ulaw to unknown
May 12 15:51:44 WARNING[14399] app_txfax.c: Unable to restore write format on 
'SIP/neopost1-8083'
May 12 15:51:44 DEBUG[14399] cdr_addon_mysql.c: cdr_mysql: inserting a CDR 
record.
May 12 15:51:44 NOTICE[14399] pbx_spool.c: Call completed to 
SIP/neopost1/0676505921
May 12 15:51:44 DEBUG[14420] app_rxfax.c: 
==
May 12 15:51:44 DEBUG[14420] app_rxfax.c: Fax successfully received.
May 12 15:51:44 DEBUG[14420] app_rxfax.c: Remote station id:
May 12 15:51:44 DEBUG[14420] app_rxfax.c: Local station id:
May 12 15:51:44 DEBUG[14420] app_rxfax.c: Pages transferred: 1
May 12 15:51:44 DEBUG[14420] app_rxfax.c: Image resolution:  7700 x 3850
May 12 15:51:44 DEBUG[14420] app_rxfax.c: Transfer Rate: 9600
May 12 15:51:44 DEBUG[14420] app_rxfax.c: 
==
May 12 15:51:44 WARNING[14420] channel.c: Unable to find a codec translation 
path from alaw to unknown
May 12 15:51:44 WARNING[14420] app_rxfax.c: Unable to restore read format on 
'mISDN/2-1'
May 12 15:51:44 WARNING[14420] channel.c: Unable to find a codec translation 
path from alaw to unknown
May 12 15:51:44 WARNING[14420] app_rxfax.c: Unable to restore write format on 
'mISDN/2-1'
May 12 15:51:44 VERBOSE[14420] logger.c:   == Spawn extension (ext-fax, in_fax, 
5) exited non-zero on 'mISDN/2-1'
May 12 15:51:44 VERBOSE[14420] logger.c: -- Executing Hangup(mISDN/2-1, 
) in new stack
May 12 15:51:44 VERBOSE[14420] logger.c:   == Spawn extension (ext-fax, h, 1) 
exited non-zero on 'mISDN/2-1'

or any idea how to overcome?
-- 
WoodOO-[P]an[G]alaktikan[A]gent-People ][ http://shadow.pganet.com
[EMAIL PROTECTED]@RedHat.users
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Re: [Asterisk-Users] Hint priority

2006-05-12 Thread Jerry Jones

I believe the hint priority must be in the same context as the phones  
extension number, in this [local]



On May 12, 2006, at 6:58 AM, richard Coco wrote:



Hi all,

i am desperating, trying to configure an OptiPoint410
with the hint priority.

Here what i have...

OptiPoint410std- exten 2001
X-Lite - exten 2002

But unfortunately no LED ON on my OptiPoint410

sip.conf
[2001]
type=friend
context=local
host=dynamic
dtmfmode=rfc2833
incominglimit=1
notifyringing=yes
subscribecontext=default
disallow=all
allow=alaw
allow=ulaw

[2002]
type=friend
context=local
host=dynamic
dtmfmode=rfc2833
incominglimit=1
notifyringing=yes
subscribecontext=default
disallow=all
allow=alaw
allow=ulaw

extensions.conf
[default]
exten = 2001,hint,SIP/2001
exten = 2002,hint,SIP/2002

[local]
exten = 2001,1,Dial(SIP/2001,10,tr)
exten = 2001,2,HangUp

exten = 2002,1,Dial(SIP/2002,10,tr)
exten = 2002,2,HangUp

Has anyone managed get hint working with an
OptiPoint4x0.

thx in advance


__
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Re: [Asterisk-Users] Re: Sangoma A200D problem

2006-05-12 Thread Dr. Michael J. Chudobiak

 Last time I had this problem was following a unclean powerdown and the 
solution was:

   - Kill Asterisk
   - Stop wanpipe
   - cd /etc/wanpipe/wan_ec
   - In there there should be 2 files:
wan_ec_pid
wan_ec_socket=
   - Delete those files
   - Perform a reboot of your system



Andre,

Thanks for the tip, it's food for thought - but I actually don't have 
those two files! Do you still have wan_ec_pid and wan_ec_socket files 
with the latest drivers (wanpipe beta 2.3.4)?


- Mike
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Re: [Asterisk-Users] email - fax gateway with billing possibilities?

2006-05-12 Thread Woodoo People .pGa!

 does anyone have an idea how it could be possible to do email - fax  
 gatewaying with asterisk + app_txfax, but still keep track of who  
 sent the fax? i've thought a little about smtp auth, but it doesn't  
 look too easy to integrate smoothly with asterisk

i don't know what your problem is.
ask the user to use a callerID as a sender ([EMAIL PROTECTED])
or pair his sender id to callerid, than do the billing on the callerid.
that's my .02
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URGENT please [Asterisk-Users] call parked / MOH

 Hello,
 
 How can I park a call or put on hold a caller from
 an
 analogue to sip agents ?
 
   PSTN===FXO/asterisk=sip agents
 
 When I press hold key or #800 the channel is hangup
 ??
 
 Harry
 Regards 












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Re: [Asterisk-Users] S100-FX v2 audio quality

2006-05-12 Thread Darrick Hartman


Tom Vile wrote:

Same problem with audio quality.  Got rid of them.  Also the context
line only allowed 12 characters and we need more than that for some
installations, I didn't want to have to rename 100 contexts to less
than 12 characters.
Which audio codecs were you using?  I'm using g729 to connect mine to 
the Asterisk box and don't have any audio problems.  Calls routed out 
over voip and my POTS lines via a TMD400 card both sounds good.  I would 
communicate the problems back to the manufacturer.  They seemed 
interested in feedback.  It is upgradeable via ftp so perhaps the 
context line can be modified by them.


I don't know how well they will perform over time.  I've only had this 
unit for 2 weeks.  We were planning on ordering several more for a 
client who wants to connect several remote users.  Both will spend 
considerable amounts of time on the phone so it has to just work.  For 
the advertised feature set, these looked impressive.


Darrick

--
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DJH Solutions, LLC
http://www.djhsolutions.com

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RE: [Asterisk-Users] Paging and Auto Answer on Grandstream GXP2000

2006-05-12 Thread Gareth Blades

There is some additional functionality coming in future firmware
versions. See
http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom

On Fri, 2006-05-12 at 14:54, Forrest Beck wrote:
 Asterisk 1.2.7.1 and Zaptel 1.2.5
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
 Sent: Thursday, May 11, 2006 6:55 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Paging and Auto Answer on Grandstream GXP2000
 
 What version of Asterisk?
 
 On 5/11/06, Forrest Beck [EMAIL PROTECTED] wrote:
 
 
 
 
  I am looking to setup paging using the auto answer feature on the
  Grandstream GXP2000.  I am thinking I will follow the method as described
  here:
 
 
 
  http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page
 
 
 
  I will setup the 4th account on the phone to auto answer.
 
 
 
  Does anyone else have a method that works better?  I also looked at the
  allpage AGI written on Voip-Info.  But it seems to dial all extensions,
 even
  the ones I don't want to use for Auto Answer.
 
 
 
  I really would like a way to group the phones instead of having them all
  listed in a dial command.
 
 
 
  exten =
  7999,2,Page(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL 
  PROTECTED]/nLocal/interal
  [EMAIL PROTECTED]|)
 
 
 
  Thanks!
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URGENT please [Asterisk-Users] call parked / MOH

  Hello,
  
  How can I park a call or put on hold a caller from
  an
  analogue to sip agents ?
  
PSTN===FXO/asterisk=sip agents
  
  When I press hold key or #800 the channel is
 hangup
  ??
  
  Harry
  Regards 














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Re: [Asterisk-Users] Paging and Auto Answer on Grandstream GXP2000


This is what I use:

[ext-paging]
exten = PAGE203,1,Set(__SIPADDHEADER=Call-Info: answer-after=0)
exten = PAGE203,n,Set(__ALERT_INFO=Ring Answer)
exten = PAGE203,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE203,n,Dial(SIP/203,5)
exten = Debug,1,Noop(dialstr is LOCAL/[EMAIL PROTECTED])
exten = 2030,1,Page(LOCAL/[EMAIL PROTECTED])

if I dial 2030 it will page extension 203.  Change accordingly.  This
works on my Grandstream and Snom phones.


On 5/12/06, Forrest Beck [EMAIL PROTECTED] wrote:

Asterisk 1.2.7.1 and Zaptel 1.2.5

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
Sent: Thursday, May 11, 2006 6:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Paging and Auto Answer on Grandstream GXP2000

What version of Asterisk?

On 5/11/06, Forrest Beck [EMAIL PROTECTED] wrote:




 I am looking to setup paging using the auto answer feature on the
 Grandstream GXP2000.  I am thinking I will follow the method as described
 here:



 http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page



 I will setup the 4th account on the phone to auto answer.



 Does anyone else have a method that works better?  I also looked at the
 allpage AGI written on Voip-Info.  But it seems to dial all extensions,
even
 the ones I don't want to use for Auto Answer.



 I really would like a way to group the phones instead of having them all
 listed in a dial command.



 exten =
 7999,2,Page(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL 
PROTECTED]/nLocal/interal
 [EMAIL PROTECTED]|)



 Thanks!
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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Re: [Asterisk-Users] S100-FX v2 audio quality


We did communicate this to the manufacturer and they fixed 1 issue
with bad power supplies.   We tried multiple codecs but it was still
unreliable, so we went back to the IAXy and no issues.  All calls came
in over a PRI.

Did not want to waste to much time on these, maybe we will look again
in the future.



On 5/12/06, Darrick Hartman [EMAIL PROTECTED] wrote:

Tom Vile wrote:
 Same problem with audio quality.  Got rid of them.  Also the context
 line only allowed 12 characters and we need more than that for some
 installations, I didn't want to have to rename 100 contexts to less
 than 12 characters.
Which audio codecs were you using?  I'm using g729 to connect mine to
the Asterisk box and don't have any audio problems.  Calls routed out
over voip and my POTS lines via a TMD400 card both sounds good.  I would
communicate the problems back to the manufacturer.  They seemed
interested in feedback.  It is upgradeable via ftp so perhaps the
context line can be modified by them.

I don't know how well they will perform over time.  I've only had this
unit for 2 weeks.  We were planning on ordering several more for a
client who wants to connect several remote users.  Both will spend
considerable amounts of time on the phone so it has to just work.  For
the advertised feature set, these looked impressive.

Darrick

--
Darrick Hartman
DJH Solutions, LLC
http://www.djhsolutions.com

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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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[Asterisk-Users] Re: Problem setting locale for voicemail

2006-05-12 Thread Álvaro Palma


Ok, this is my voicemail.conf:

[general]
attach=yes
charset=ISO-8859-1
emailbody=${VM_NAME}:\n\nUd ha recibido un nuevo mensaje de voz, de 
${VM_DUR} segundos.\n\nTeléfono: ${VM_CALLERID}.\nFecha: 
${VM_DATE}.\n\nEl mensaje ha sido adjunto a este correo.\n\nPor favor, 
no responda este mensaje, esta casilla no es válida.

emaildateformat=%A %d de %B de %Y, %T
emailsubject=Nuevo mensaje de voz para ${VM_MAILBOX}
envelope=yes
format=wav49
fromstring=Correo de Voz
language=es
maxgreet=60
maxlogins=3
maxmessage=180
maxmsg=100
maxsilence=10
minmessage=3
nextaftercmd=yes
operator=no
pbxskip=yes
review=no
saycid=yes
sendvoicemail=yes
[EMAIL PROTECTED]
skipms=3000
silencethreshold=128
sayduration=yes
saydurationm=2

[zonemessages]
chile_continental=Chile/Continental|'vm-received' 'digits/at' A d B R
chile_insular=Chile/EasterIsland|'vm-received' 'digits/at' A d B R

[default]

01 = pass01,Test01,[EMAIL PROTECTED],,tz=chile_continental|delete=yes

I don't know is language is an option for voicemail.conf. However, in 
zapata.conf and sip.conf (the two types of channels I'm using), langauge 
it is an option. In both, I set it to language=es. In fact, I 
downloaded sounds for spanish, and I can hear all the prompts correctly 
in spanish, but the email text is still localized as english. I tried 
with my locale set as es, es_ES and es_CL, but nothing happened.


Thanks for your help

 Hi,

 If u write how you set the voice mail, its very useful for me and
 users also. Please write step by step procedure to implement voice
 mail in asterisk.

 Thanks  Regards,
 Chandra.

Álvaro Palma [EMAIL PROTECTED] wrote:

 I've set voicemail almost successfully, only a minor detail remains
 :-)
 I can't get the dates in my local language (spanish). In sip.conf,
 zapata.conf and voicemail.conf, I've set:

 language=es

 and my locale is es also. However, the days and months names still
 appear in english in the emails!!!

 Thursday 11 de May de 2006, 18:49:34.

 instead of

 Martes 11 de mayo de 2006, 18:49:34.

 Anybody knows a fix for it?

 Thanks a lot for your help.

--
Atly.
Álvaro Palma
OPS Ingeniería Ltda.
F: (56 2)5805905
Skype: alvaro_palma_aste
MSN  : [EMAIL PROTECTED]
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Re: URGENT please [Asterisk-Users] call parked / MOH

2006-05-12 Thread Alex Robar

Harry,Please note that you have sent this message to the group several times today. If anyone has an answer for you, they will reply. There is no need to continually send it to the group.Alex
On 5/12/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
  Hello,   How can I park a call or put on hold a caller from  an  analogue to sip agents ?   PSTN===FXO/asterisk=sip agents 
  When I press hold key or #800 the channel is hangup  ??   Harry  Regards___
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Re: [Asterisk-Users] Bristuffed Asterisk: Hangup problems

2006-05-12 Thread Steve Davies


On 5/12/06, stoffell [EMAIL PROTECTED] wrote:

On 5/11/06, Tim Robinson [EMAIL PROTECTED] wrote:
 There is a lot of junk in your zapata.conf that you do not need, as it
 relates to analogue lines.  This might be causing confusion?

I have tried a similary config to yours, doesn't helps. I haven't got
this problem on an E1, just on the newer bristuff'd packages.

I have sent an email to junghanns.net about this, but haven't received
an answer yet. If I do receive anything, I'll post it back to the
thread.

If there are any other things I might check, please let me know..


I am still using the older 0.2.0 bristuff packages, and found that the
April RC8r release causes more issues than it solves. Particularly in
the 'qozap' driver, which has become unusable on a quad card that does
not have all 4 lines connected... The patches to core asterisk on the
other hand are pretty good. Some of the backported snom/SIP patches in
particular.

I have similarly never had a response to an email on the subject even
though we resell a reasonable number of their quad and single port
cards.

My current favourite version is RC8r with the qozap source rolled back
to RC8n, but that is only helpful to 1.0.x systems - Perhaps there is
some mileage in you doing a similar thing with the 0.3.0-pre branch.
Roll back as far as necessary in the pre branch so that the hangup
problem is solved, and then do a diff and see if the change that
causes the issue is obvious?

Cheers,
Steve
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Re: [Asterisk-Users] Hint priority

2006-05-12 Thread Steve Davies


On 5/12/06, Jerry Jones [EMAIL PROTECTED] wrote:

I believe the hint priority must be in the same context as the phones
extension number, in this [local]



Additionally, it may not be the first 'exten =' line, at least in
some versions, so best to put them at the end of the context.

PLUS: Avoid SIP registrations with a minus '-' in them as this breaks
on several versions.

Hope that helps,
Steve
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RE: [Asterisk-Users] features.conf *1 Call Recording




It's quite strange. When I press *1 I do not hear a tone 
indicated that it's even trying to record.

David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodatasolutions.com

Tel: (519) 963-3020
Fax: (519) 451-6615

 Lead, follow or get out of the 
way! 


This message has originated from Autodata Solutions. The attached material is 
the Confidential and Proprietary Information of Autodata Solutions. This email 
and any files transmitted with it are confidential and intended solely for the 
use of the individual or entity to whom they are addressed. If you have received 
this email in error please delete this message and notify the Autodata system 
administrator at [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED]



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Dave 
MorrowSent: Friday, May 12, 2006 8:39 AMTo: Asterisk Users 
Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] 
features.conf *1 Call Recording

Yes. I did.

David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodatasolutions.com

Tel: (519) 963-3020
Fax: (519) 451-6615

 Lead, follow or get out of the 
way! 


This message has originated from Autodata Solutions. The attached material is 
the Confidential and Proprietary Information of Autodata Solutions. This email 
and any files transmitted with it are confidential and intended solely for the 
use of the individual or entity to whom they are addressed. If you have received 
this email in error please delete this message and notify the Autodata system 
administrator at [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED]



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Giridhar 
Reddy BandiSent: Friday, May 12, 2006 3:41 AMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] features.conf *1 Call Recording
did you include automon = *1 in your features.conf ?? it 
should be somthing like this [featuremap]automon = 
*1 
--Giridhar Bandi 
On 5/12/06, Dave 
Morrow [EMAIL PROTECTED] 
wrote: 
Thanks 
  for the response.How would I change the DTMF transfer 
  mode?David MorrowTechnical Systems LeadAutodata Solutions 
  Company[EMAIL PROTECTED] http://www.autodatasolutions.comTel: 
  (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of 
  the way! This message has originated from Autodata Solutions. The 
  attached material is the Confidential and Proprietary Information of 
  AutodataSolutions. This email and any files transmitted with it are 
  confidentialand intended solely for the use of the individual or entity to 
  whom they are addressed. If you have received this email in error please 
  deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] 
  mailto: [EMAIL PROTECTED]-Original 
  Message-From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]] 
  On Behalf Of FabioSent: Thursday, May 11, 2006 6:55 PMTo: Asterisk 
  Users Mailing List - Non-Commercial Discussion Subject: RE: 
  [Asterisk-Users] features.conf *1 Call Recordingif you ar using SIP 
  clients, try changing DTMF transfer mode.For test use sip 
  debugon your * console, then place a call and watch the output. In INFO or 
  rfc2833 mode you must see the codes like SIP messages. If you are 
  usinginband transfer mode (DTMF codes aretransferred like 
  sounds) you don'tsee the codes.Also, try adjusting 
  featuredigittimeout in features.conf :[general]featuredigittimeout 
  = 2000 ; 2 secondsbecause the default 500ms is a very short 
  time.Fabio Olaechea3Tech SRLCalle 48 Nro 632, Of. 
  67.La Plata, CP B1900AMZBuenos Aires, Argentina. Tel. +54 221 445 
  0244 Ext. 301Fax. +54 221 445 0245www.trestech.com.ar-Mensaje 
  original-De: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]]En 
  nombre de Dave MorrowEnviado el: Miercoles, 10 de Mayo de 2006 02:48 
  p.m.Para: Asterisk Users Mailing List - Non-Commercial 
  DiscussionAsunto: RE: [Asterisk-Users] features.conf *1 Call 
  RecordingOK. You lost me.David MorrowTechnical 
  Systems LeadAutodata Solutions Company [EMAIL PROTECTED]http://www.autodatasolutions.comTel: 
  (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of 
  the way!  This message has originated from Autodata Solutions. The 
  attachedmaterial is the Confidential and Proprietary Information of 
  AutodataSolutions. This email and any files transmitted with it are 
  confidential and intended solely for the use of the individual or entity 
  to whom theyare addressed. If you have received this email in error please 
  deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] 
  mailto:[EMAIL PROTECTED]-Original 
  Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] 
  On Behalf Of AlejandroVargasSent: Wednesday, May 10, 2006 10:29 AM 
  To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: 
  Re: [Asterisk-Users] features.conf *1 Call

[Asterisk-Users] Voicemail WAV to PDA Problems

2006-05-12 Thread Peder @ NetworkOblivion

Our asterisk server has been up and running for over a year and it works 
great.  I have emails going to my account as an attachment and I can 
listen to them on the desktop and it works fine.  I just got a T-Mobile 
MDA that runs Windows Pocket (or whatever they call it) and it can check 
email.  If I have it download the email, it gets the attachment, but it 
can't seem to play it (it CAN play wav files).  If I take the email that 
was sent to my home account and then forward it to myself and let the 
MDA pick it up, then it can play the attachment.  So clearly it isn't an 
issue playing WAV's, or even WAV's from Asterisk, it's some email 
attachment issue with the way Asterisk or Postfix sends the attachment. 
 Has anybody else run into this problem?  If so, any help would be 
appreciated.


Peder

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[Asterisk-Users] Music on Hold restart at beginning for each call

2006-05-12 Thread Tim Sharp

I am using the m option on the dial command to play a message instead of 
ringing.  The message is something like please wait while I try to locate your 
party so I need it to start at the beginning for each call.  I think there 
might be a way in 1.2.x be we are not ready to upgrade yet so a solution for 
1.0.9 is what I am after.  Thanks.  
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RE: [Asterisk-Users] Voicemail WAV to PDA Problems

2006-05-12 Thread Kerry Garrison

Our system is running all of the latest code and freepbx and would send the
attachment to my MDA just fine and I was able to play it without any
problem. My problem was that the MDA is a worthless turd and a complete joke
as a phone. I took it back and switched to the backberry 8700g which has its
own attachment problems. 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Peder @ NetworkOblivion
 Sent: Friday, May 12, 2006 9:02 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Voicemail WAV to PDA Problems
 
 Our asterisk server has been up and running for over a year 
 and it works great.  I have emails going to my account as an 
 attachment and I can listen to them on the desktop and it 
 works fine.  I just got a T-Mobile MDA that runs Windows 
 Pocket (or whatever they call it) and it can check email.  If 
 I have it download the email, it gets the attachment, but it 
 can't seem to play it (it CAN play wav files).  If I take the 
 email that was sent to my home account and then forward it 
 to myself and let the MDA pick it up, then it can play the 
 attachment.  So clearly it isn't an issue playing WAV's, or 
 even WAV's from Asterisk, it's some email attachment issue 
 with the way Asterisk or Postfix sends the attachment. 
   Has anybody else run into this problem?  If so, any help 
 would be appreciated.
 
 Peder
 
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[Asterisk-Users] Help Avaya 4606

2006-05-12 Thread Carlos Rojas

Hello all,

I have asterisk working well with, Sipura, but I do not manage to form
several phones avaya 4606, someone could have formed one avaya with
asterisk?

is it possible?

update the firmware of the phone, but I do not achieve that it registers, 


I hope that someone could help me

greetings to all

Carlos Rojas
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RE: [Asterisk-Users] MeetME Conferencing

2006-05-12 Thread Josh McAllister

Your welcome. It certainly could be done entirely in the dialplan using similar 
logic, but this required a bit less mental horsepower. If your desire to avoid 
AGI, is based on performance concerns, note that I have systems (Dell 2850 
2xXEON 3.0) that terminate 8 PRIs and have had ALL channels loaded up with perl 
AGI scripts and never skipped a beat. FWIW, these servers have 4G ram, and run 
64bit RHES. Either way, glad I could get you closer to the end. 

Josh McAllister

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Damon Estep
 Sent: Friday, May 12, 2006 7:06 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] MeetME Conferencing
 
 Josh,
 
 Thank you!
 
 I think the AGI could be bypassed by doing a realtime() to get the PIN
 from mySQL, also returning the variable that defines admin or user and
 jumping in the dialplan accordingly. Otherwise I would just end up having
 the AGI do the query because there is a need to store the users in the
 database to facilitate easy management.
 
 The admin menu and marked user options seem to be the key to making this
 work, so I will play with those.
 
 
 
 
 
 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Josh McAllister
 Sent: Friday, May 12, 2006 2:08 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] MeetME Conferencing
 
 Ok, the script below (meetme.agi) will prompt for a valid pin up to 3
 times. If the pin matches one of the defined Admin pins, it will set the
 dialplan priority to 10 and exit, if User, sets to 20 and exits. Otherwise
 Hangs up.
 
 In the case of admin, these MeetMe options are used:
 a - Admin mode
 A - Marked mode
 c - Announce number of participants (optional of course)
 s - Present Admin menu by pressing '*'
 x - close conf when last marked user leaves.
 
 In the case of user:
 c s x are used as above, but we add:
 w - wait until marked user enters. (Plays MoH until then)
 
 The dialplan assumes you have a static pinless conference setup as conf
 #10.
 
 extensions.conf:
 exten = 5552323,1,Wait(1)
 exten = 5552323,2,Answer()
 exten = 5552323,3,AGI(meetme.agi)
 exten = 5552323,4,NoOp(Invalid Pin)
 exten = 5552323,5,Hangup()
 
 exten = 5552323,10,NoOp(Admin Pin)
 exten = 5552323,11,MeetMe(10,aAcsx)
 exten = 5552323,12,Hangup()
 
 exten = 5552323,20,NoOp(User Pin)
 exten = 5552323,21,MeetMe(10,cswx)
 exten = 5552323,22,Hangup()
 
 
 
 The script of course requires the Asterisk::AGI module.
 
 meetme.agi:
 
 #!/usr/bin/perl
 use Asterisk::AGI;
 my $AGI = new Asterisk::AGI;
 my $input = { %{$AGI-ReadParse()} };
 
 #our $DEBUG = 1;
 
 my @UserPins = ('1','2');
 my @AdminPins = ('9','8');
 
 my $mode = collectPin($AGI,5);
 
 $AGI-verbose(collectPin got '$mode') if $DEBUG;
 
 if ($mode eq 'Admin') {
$AGI-set_priority(10);
 } elsif ($mode eq 'User') {
$AGI-set_priority(20);
 } else {
$AGI-stream_file(goodbye,'');
$AGI-hangup;
 }
 
 exit;
 
 sub collectPin {
my $AGI = shift;
my $maxdigits = shift;
 
my $tries = 0;
 
#Three tries to select an existing pin.
while ($tries  3) {
   $AGI-stream_file(please-try-again,'') if $tries  0;
   $tries++;
   my $pin = $AGI-get_data('enter-conf-pin-number', 1,
 $maxdigits);
   $AGI-verbose(Got PIN $pin.) if $DEBUG;
   next unless $pin  0;
 
   if ( grep(/^$pin$/, @AdminPins) ) {
  $AGI-stream_file(pin-number-accepted,'');
  return 'Admin';
   } elsif ( grep(/^$pin$/, @UserPins) ) {
  $AGI-stream_file(pin-number-accepted,'');
  return 'User';
   } else {
  $AGI-stream_file(conf-invalidpin,'');
   }
}
 
return undef;
 }
 
 
 What can I say, I was bored.
 
 Enjoy,
 
 Josh McAllister
 
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Damon Estep
 Sent: Thursday, May 11, 2006 10:37 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] MeetME Conferencing
 
 Static configs for the conference rooms are not an issue.
 
 The main goal is to allow the moderator to determine when the conference
 “starts” by having all participants hearing MOH until the moderator starts
 the interactive call with a  PIN known only to the moderator, and then
 allowing the moderator (and only the moderator) to kick out all users from
 the keypad when the call is over.
 
 An additional benefit would be gained if authenticate() or realtime() app
 commands could be used against a mysql database for the participant and
 moderator pins so an app could be written easily to allow changing of the
 PINS in the database.
 
 
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Dan Austin
 Sent: Thursday, May 11, 2006 10:29 PM

[Asterisk-Users] Cisco 7970 problems

2006-05-12 Thread Hall, Eric M.




Has anyone had problems with a Cisco 7970 running sip image 
SIP70.8.0-2SR1S hanging up zap channels?

Calls to SIP and IAX 
are fine. Just when the call goes out via the zap channels

I have some Cisco 
7960 running SIP and they work fine.

Any 
ideas?

Thanks-Eric Hall

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Re: [Asterisk-Users] features.conf *1 Call Recording

2006-05-12 Thread Giridhar Reddy Bandi

hi Dave i get the following log on *CLI   -- Attempting native bridge of SIP/200-39f4 and SIP/204-2ce4 -- Playing 'beep' (language 'en') -- User hit '*1' to record call. filename: wav|auto-1147452537-200-204|m
 -- Playing 'beep' (language 'en') -- User hit '*1' to stop recording call. -- Attempting native bridge of SIP/200-39f4 and SIP/204-2ce4what are you using as SIP client ( imean softphone/ analog phone + ATA / IPphone ) ?
if you are using a softphone and that doesnot have a dtmf signaling then asterisk will not be able to recognize that you are pressing.--Giridhar Bandi On 5/12/06, 
Dave Morrow [EMAIL PROTECTED] wrote:





It's quite strange. When I press *1 I do not hear a tone 
indicated that it's even trying to record.

David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodatasolutions.com

Tel: (519) 963-3020
Fax: (519) 451-6615

 Lead, follow or get out of the 
way! 


This message has originated from Autodata Solutions. The attached material is 
the Confidential and Proprietary Information of Autodata Solutions. This email 
and any files transmitted with it are confidential and intended solely for the 
use of the individual or entity to whom they are addressed. If you have received 
this email in error please delete this message and notify the Autodata system 
administrator at [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED]




From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]] On Behalf Of Dave 
MorrowSent: Friday, May 12, 2006 8:39 AMTo: Asterisk Users 
Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] 
features.conf *1 Call Recording

Yes. I did.

David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodatasolutions.com

Tel: (519) 963-3020
Fax: (519) 451-6615

 Lead, follow or get out of the 
way! 


This message has originated from Autodata Solutions. The attached material is 
the Confidential and Proprietary Information of Autodata Solutions. This email 
and any files transmitted with it are confidential and intended solely for the 
use of the individual or entity to whom they are addressed. If you have received 
this email in error please delete this message and notify the Autodata system 
administrator at [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED]




From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]] On Behalf Of Giridhar 
Reddy BandiSent: Friday, May 12, 2006 3:41 AMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] features.conf *1 Call Recording
did you include automon = *1 in your features.conf ?? it 
should be somthing like this [featuremap]automon = 
*1 
--Giridhar Bandi 
On 5/12/06, Dave 
Morrow [EMAIL PROTECTED] 
wrote: 
Thanks 
  for the response.How would I change the DTMF transfer 
  mode?David MorrowTechnical Systems LeadAutodata Solutions 
  Company[EMAIL PROTECTED] 
http://www.autodatasolutions.comTel: 
  (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of 
  the way! This message has originated from Autodata Solutions. The 
  attached material is the Confidential and Proprietary Information of 
  AutodataSolutions. This email and any files transmitted with it are 
  confidentialand intended solely for the use of the individual or entity to 
  whom they are addressed. If you have received this email in error please 
  deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED]
 
  mailto: [EMAIL PROTECTED]-Original 
  Message-From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]] 
  On Behalf Of FabioSent: Thursday, May 11, 2006 6:55 PMTo: Asterisk 
  Users Mailing List - Non-Commercial Discussion Subject: RE: 
  [Asterisk-Users] features.conf *1 Call Recordingif you ar using SIP 
  clients, try changing DTMF transfer mode.For test use sip 
  debugon your * console, then place a call and watch the output. In INFO or 
  rfc2833 mode you must see the codes like SIP messages. If you are 
  usinginband transfer mode (DTMF codes aretransferred like 
  sounds) you don'tsee the codes.Also, try adjusting 
  featuredigittimeout in features.conf :[general]featuredigittimeout 
  = 2000 ; 2 secondsbecause the default 500ms is a very short 
  time.Fabio Olaechea3Tech SRLCalle 48 Nro 632, Of. 
  67.La Plata, CP B1900AMZBuenos Aires, Argentina. Tel. +54 221 445 
  0244 Ext. 301Fax. +54 221 445 0245www.trestech.com.ar-Mensaje 
  original-De: [EMAIL PROTECTED][mailto:
[EMAIL PROTECTED]]En 
  nombre de Dave MorrowEnviado el: Miercoles, 10 de Mayo de 2006 02:48 
  p.m.Para: Asterisk Users Mailing List - Non-Commercial 
  DiscussionAsunto: RE: [Asterisk-Users] features.conf *1 Call 
  RecordingOK. You lost me.David MorrowTechnical 
  Systems LeadAutodata Solutions Company [EMAIL PROTECTED]
http://www.autodatasolutions.comTel: 
  (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of 
  the way!  This message has originated from Autodata Solutions. The 
  attachedmaterial is the Confidential and

RE: [Asterisk-Users] features.conf *1 Call Recording




I have one Sipura SPA-841 which is configured to use 
dtmfmode=info and one Cisco 7905 which is using the default signalling (I 
believe this is rfc2833) 
I have also set relaxdtmf=yes in 
sip.conf

I've tried pressing *1 on both phones (they are both on my 
desk) and both behave the same.

;; Sample Parking 
configuration;

[general]parkext = 
700 
; What ext. to dial to parkparkpos = 
701-720 
; What extensions to park calls oncontext = 
parkedcalls ; Which 
context parked calls are in;parkingtime = 
45 
; Number of seconds a call can be parked 
for 
; (default is 45 seconds);transferdigittimeout = 
3 ; Number of seconds to wait between digits when 
transfering a call;courtesytone = 
beep ; Sound 
file to play to the parked 
caller 
; when someone dials a parked call;xfersound = 
beep 
; to indicate an attended transfer is complete;xferfailsound = 
beeperr ; to indicate a failed 
transfer;adsipark = 
yes 
; if you want ADSI parking announcements;findslot = 
next 
; Continue to the 'next' parking space. Defaults to 'first' 
available;pickupexten = 
*8 
; Configure the pickup extension. Default is *8featuredigittimeout = 
2000 ; Max time (ms) between digits 
for 
; feature activation. Default is 500

[featuremap]blindxfer = 
#1 ; Blind 
transferdisconnect = 
*0 
; Disconnectautomon = 
*1 
; One Touch Recordatxfer = 
*2 
; Attended transfer

[applicationmap];testfeature = 
#9,callee,Playback,tt-monkeys ;Play tt-monkes 
to 
;callee if #9 was pressed

~~~

David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodatasolutions.com

Tel: (519) 963-3020
Fax: (519) 451-6615

 Lead, follow or get out of the 
way! 


This message has originated from Autodata Solutions. The attached material is 
the Confidential and Proprietary Information of Autodata Solutions. This email 
and any files transmitted with it are confidential and intended solely for the 
use of the individual or entity to whom they are addressed. If you have received 
this email in error please delete this message and notify the Autodata system 
administrator at [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED]



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Giridhar 
Reddy BandiSent: Friday, May 12, 2006 12:55 PMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] features.conf *1 Call Recording
hi Dave i get the following log on *CLI 
 -- Attempting native bridge of SIP/200-39f4 and 
SIP/204-2ce4 -- Playing 'beep' (language 
'en') -- User hit '*1' to record call. filename: 
wav|auto-1147452537-200-204|m  -- Playing 'beep' (language 
'en') -- User hit '*1' to stop recording 
call. -- Attempting native bridge of SIP/200-39f4 and 
SIP/204-2ce4what are you using as SIP client ( imean softphone/ analog 
phone + ATA / IPphone ) ? if you are using a softphone and that doesnot 
have a dtmf signaling then asterisk will not be able to recognize that you 
are pressing.--Giridhar Bandi 
On 5/12/06, Dave 
Morrow [EMAIL PROTECTED] 
wrote:

  
  
  It's quite 
  strange. When I press *1 I do not hear a tone indicated that it's even trying 
  to record.
  
  
  David Morrow
  Technical Systems Lead
  Autodata Solutions 
Company
  [EMAIL PROTECTED]
  http://www.autodatasolutions.com
  
  Tel: (519) 963-3020
  Fax: (519) 451-6615
  
   Lead, follow or get out of 
  the way! 
  
  
  This message has originated from Autodata Solutions. The attached material 
  is the Confidential and Proprietary Information of Autodata Solutions. This 
  email and any files transmitted with it are confidential and intended solely 
  for the use of the individual or entity to whom they are addressed. If you 
  have received this email in error please delete this message and notify the 
  Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 
  
  
  
  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of 
  Dave MorrowSent: Friday, May 12, 2006 8:39 AMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [Asterisk-Users] features.conf *1 Call Recording
  
  Yes. I 
  did.
  
  David Morrow
  Technical Systems Lead
  Autodata Solutions 
Company
  [EMAIL PROTECTED]
  http://www.autodatasolutions.com
  
  Tel: (519) 963-3020
  Fax: (519) 451-6615
  
   Lead, follow or get out of 
  the way! 
  
  
  This message has originated from Autodata Solutions. The 
  attached material is the Confidential and Proprietary Information of Autodata 
  Solutions. This email and any files transmitted with it are confidential and 
  intended solely for the use of the individual or entity to whom they are 
  addressed. If you have received this email in error please delete this message 
  and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 
  
  
  
  From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of 
  Giridhar Reddy BandiSent: Friday, May 12, 2006 3:41 
  AMTo: Asterisk Users Mailing List -

[Asterisk-Users] Having Rinback tone generation issues with 1.2.7.1

2006-05-12 Thread Alberto Sagredo

Today i move our central server to 1.2.7.1 , and im having some issues 
with SPA Phones and RinbackTone. Without r option, it also happens. Is 
having anyone this issue? I think it has not been changed anything 
sustancially to happen this to me.


It is happening between extensiones (canreinvite=yes), outbount trunking 
(no gets rinback tone (generated by phone).


Regards


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[Asterisk-Users] RE: snmp and asterisk

hi david,

can you explain me this please?
If Sangoma hardware support snmp ithink it would be a
better choice than digium .

How can we know the state of the sangoma cards with an
snmp agent ?

Harry
--- David Yat Sin [EMAIL PROTECTED] a écrit :

 Hi Harry,
 The Sangoma Card when used for TDM Voice will work
 under zaptel, so you
 would need to perform the SNMP through Asterisk.
 
 Regards,
 David Yat Sin
 Sangoma Technologies
 (905) 474 1990 x119
 (800) 388 2475 x199
 MSN: [EMAIL PROTECTED]
 Email: [EMAIL PROTECTED]
 Wiki: http://sangoma.editme.com
  
 
 
  
  -Original Message-
  From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED]
  Sent: Friday, May 12, 2006 3:42 AM
  To: [EMAIL PROTECTED]
  Subject: ***SANGOMA INFORMATION REQUEST***
  
  CSV for Maximzer:
  -
  e-citel harry   +33 493 450 084 cannes  
  france
  [EMAIL PROTECTED]   How sangoma support snmp
 with asterisk 
  pbx ?   Info
  request from web
  -
  harry has requested information on Fri, 12 May
 2006 03:42:29 EDT.
  
  Primary Address
  ---
  Company:e-citel
  Contact:harry
  Email:  [EMAIL PROTECTED]
  Phone:  +33 493 450 084
  City:   cannes
  Country:france
  -
  Product Interest:   
  -
  How did you hear of Sangoma:Keywords: 
  -
  Additional Comments:
  How sangoma support snmp with asterisk pbx ?
  
  
  
  
 
 
 









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RE: [Asterisk-Users] Voicemail WAV to PDA Problems

2006-05-12 Thread Dean Collins

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Kerry Garrison
 Sent: Friday, 12 May 2006 12:07 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] Voicemail WAV to PDA Problems

 Our system is running all of the latest code and freepbx and would
send the
 attachment to my MDA just fine and I was able to play it without any
 problem. My problem was that the MDA is a worthless turd and a
complete joke
 as a phone. I took it back and switched to the backberry 8700g which
has its
 own attachment problems.

  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of
  Peder @ NetworkOblivion
  Sent: Friday, May 12, 2006 9:02 AM
  To: asterisk-users@lists.digium.com
  Subject: [Asterisk-Users] Voicemail WAV to PDA Problems

  Our asterisk server has been up and running for over a year
  and it works great.  I have emails going to my account as an
  attachment and I can listen to them on the desktop and it
  works fine.  I just got a T-Mobile MDA that runs Windows
  Pocket (or whatever they call it) and it can check email.  If
  I have it download the email, it gets the attachment, but it
  can't seem to play it (it CAN play wav files).  If I take the
  email that was sent to my home account and then forward it
  to myself and let the MDA pick it up, then it can play the
  attachment.  So clearly it isn't an issue playing WAV's, or
  even WAV's from Asterisk, it's some email attachment issue
  with the way Asterisk or Postfix sends the attachment.
Has anybody else run into this problem?  If so, any help
  would be appreciated.

  Peder

Anyone who is going to be visiting JavaOne in San Francisco next week
come and say hi to me on the Savaje Stand (www.savaje.com).

Lets just say we have something interesting to show you that will solve
this problem (and more).

Cheers,

Dean

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RE: [Asterisk-Users] Paging and Auto Answer on Grandstream GXP2000

2006-05-12 Thread Forrest Beck

Thanks.  I like that method. 

Do you think if I add all my extensions (say 40 of them) to new Dial
commands after exten = PAGE203,n,Dial(SIP/203,5)

Like this:
exten = PAGE203,n,Dial(SIP/203,5)
exten = PAGE203,n,Dial(SIP/204,5)
exten = PAGE203,n,Dial(SIP/205,5)
exten = PAGE203,n,Dial(SIP/206,5)

will work?

The only think I am not sure about is if a phone doesn't answer in the 5
seconds allowed, it will hang until that 5 seconds is up to parse the next
line.

Also, if it is dialing each phone line by line will there be a delay for all
the phones are dialed and pick up.

Thanks again.

Forrest

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
Sent: Friday, May 12, 2006 10:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Paging and Auto Answer on Grandstream GXP2000

This is what I use:

[ext-paging]
exten = PAGE203,1,Set(__SIPADDHEADER=Call-Info: answer-after=0)
exten = PAGE203,n,Set(__ALERT_INFO=Ring Answer)
exten = PAGE203,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten = PAGE203,n,Dial(SIP/203,5)
exten = Debug,1,Noop(dialstr is LOCAL/[EMAIL PROTECTED])
exten = 2030,1,Page(LOCAL/[EMAIL PROTECTED])

if I dial 2030 it will page extension 203.  Change accordingly.  This
works on my Grandstream and Snom phones.


On 5/12/06, Forrest Beck [EMAIL PROTECTED] wrote:
 Asterisk 1.2.7.1 and Zaptel 1.2.5

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
 Sent: Thursday, May 11, 2006 6:55 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Paging and Auto Answer on Grandstream
GXP2000

 What version of Asterisk?

 On 5/11/06, Forrest Beck [EMAIL PROTECTED] wrote:
 
 
 
 
  I am looking to setup paging using the auto answer feature on the
  Grandstream GXP2000.  I am thinking I will follow the method as
described
  here:
 
 
 
  http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page
 
 
 
  I will setup the 4th account on the phone to auto answer.
 
 
 
  Does anyone else have a method that works better?  I also looked at the
  allpage AGI written on Voip-Info.  But it seems to dial all extensions,
 even
  the ones I don't want to use for Auto Answer.
 
 
 
  I really would like a way to group the phones instead of having them all
  listed in a dial command.
 
 
 
  exten =
  7999,2,Page(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL 
  PROTECTED]/nLocal/interal
  [EMAIL PROTECTED]|)
 
 
 
  Thanks!
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 --
 Tom Vile
 Baldwin Technology Solutions, Inc
 Consulting - Web Design - VoIP Telephony
 www.baldwintechsolutions.com
 Phone: 518-631-2855 x205
 Fax: 518-631-2856
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-- 
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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[Asterisk-Users] Dial Command Reference for SIP channel




Hi all. I was 
reading a sample config someone had posted relating to call forwarding, and in 
it, they use a Dial command with components that I cannot find any reference 
to.

Can someone point me 
to a reference which could explain the difference between 
Dial(SIP/100|20|Ttr,,wW) and Dial(SIP/100,,wW) 
Specifically, what 
is the |20|Ttr ? I cannot seem to find any reference which would indicate 
this is even a valid format for the SIP channel.

David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodatasolutions.com

Tel: (519) 963-3020
Fax: (519) 451-6615

 Lead, follow or get out of the 
way! 


This message has originated from Autodata Solutions. The attached material is 
the Confidential and Proprietary Information of Autodata Solutions. This email 
and any files transmitted with it are confidential and intended solely for the 
use of the individual or entity to whom they are addressed. If you have received 
this email in error please delete this message and notify the Autodata system 
administrator at [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED]

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RE: [Asterisk-Users] features.conf *1 Call Recording




I found the issue.

It was my Dial command!

In my dialplan I had Dial(SIP/100|20|Ttr,,wW) as this was 
something I gleaned from a sample config for call forwarding. I removed 
the |20|Ttr andnow the call recording works! Anyone know what the 
|20|Ttr did anyhow?

David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodatasolutions.com

Tel: (519) 963-3020
Fax: (519) 451-6615

 Lead, follow or get out of the 
way! 


This message has originated from Autodata Solutions. The attached material is 
the Confidential and Proprietary Information of Autodata Solutions. This email 
and any files transmitted with it are confidential and intended solely for the 
use of the individual or entity to whom they are addressed. If you have received 
this email in error please delete this message and notify the Autodata system 
administrator at [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED]



From: Dave Morrow Sent: Friday, May 
12, 2006 10:41 AMTo: 'Asterisk Users Mailing List - Non-Commercial 
Discussion'Subject: RE: [Asterisk-Users] features.conf *1 Call 
Recording

It's quite strange. When I press *1 I do not hear a tone 
indicated that it's even trying to record.

David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodatasolutions.com

Tel: (519) 963-3020
Fax: (519) 451-6615

 Lead, follow or get out of the 
way! 


This message has originated from Autodata Solutions. The attached material is 
the Confidential and Proprietary Information of Autodata Solutions. This email 
and any files transmitted with it are confidential and intended solely for the 
use of the individual or entity to whom they are addressed. If you have received 
this email in error please delete this message and notify the Autodata system 
administrator at [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED]



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Dave 
MorrowSent: Friday, May 12, 2006 8:39 AMTo: Asterisk Users 
Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] 
features.conf *1 Call Recording

Yes. I did.

David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodatasolutions.com

Tel: (519) 963-3020
Fax: (519) 451-6615

 Lead, follow or get out of the 
way! 


This message has originated from Autodata Solutions. The attached material is 
the Confidential and Proprietary Information of Autodata Solutions. This email 
and any files transmitted with it are confidential and intended solely for the 
use of the individual or entity to whom they are addressed. If you have received 
this email in error please delete this message and notify the Autodata system 
administrator at [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED]



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Giridhar 
Reddy BandiSent: Friday, May 12, 2006 3:41 AMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] features.conf *1 Call Recording
did you include automon = *1 in your features.conf ?? it 
should be somthing like this [featuremap]automon = 
*1 
--Giridhar Bandi 
On 5/12/06, Dave 
Morrow [EMAIL PROTECTED] 
wrote: 
Thanks 
  for the response.How would I change the DTMF transfer 
  mode?David MorrowTechnical Systems LeadAutodata Solutions 
  Company[EMAIL PROTECTED] http://www.autodatasolutions.comTel: 
  (519) 963-3020Fax: (519) 451-6615 Lead, follow or get out of 
  the way! This message has originated from Autodata Solutions. The 
  attached material is the Confidential and Proprietary Information of 
  AutodataSolutions. This email and any files transmitted with it are 
  confidentialand intended solely for the use of the individual or entity to 
  whom they are addressed. If you have received this email in error please 
  deletethis message and notify the Autodata system administrator at[EMAIL PROTECTED] 
  mailto: [EMAIL PROTECTED]-Original 
  Message-From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]] 
  On Behalf Of FabioSent: Thursday, May 11, 2006 6:55 PMTo: Asterisk 
  Users Mailing List - Non-Commercial Discussion Subject: RE: 
  [Asterisk-Users] features.conf *1 Call Recordingif you ar using SIP 
  clients, try changing DTMF transfer mode.For test use sip 
  debugon your * console, then place a call and watch the output. In INFO or 
  rfc2833 mode you must see the codes like SIP messages. If you are 
  usinginband transfer mode (DTMF codes aretransferred like 
  sounds) you don'tsee the codes.Also, try adjusting 
  featuredigittimeout in features.conf :[general]featuredigittimeout 
  = 2000 ; 2 secondsbecause the default 500ms is a very short 
  time.Fabio Olaechea3Tech SRLCalle 48 Nro 632, Of. 
  67.La Plata, CP B1900AMZBuenos Aires, Argentina. Tel. +54 221 445 
  0244 Ext. 301Fax. +54 221 445 0245www.trestech.com.ar-Mensaje 
  original-De: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]]En 
  nombre de Dave MorrowEnviado el: Miercoles, 10 de Mayo

Re: [Asterisk-Users] Paging and Auto Answer on Grandstream GXP2000