Re: [asterisk-users] Inaccurate CDRs

[Dumpolid Exeplish]

This is an update on the issues of CDR inaccuracies, i hope this will help someone in need. 
In order to remove the Authenticate() function and still be able to perform call accounting and authentication, we pass the authetication process through an AGI. the reason beig that the Authenticate() application always awnsers the channel first before authentication. This will cause irregularities in terms of billing since the user will have to pay for both the 'authentication' time and the call processing time. My authentication Macros calls this perl script to enable authentication without awnsering the channel.

 
 
 
#!/usr/bin/perl
use Asterisk::AGI;open (PIN1,'/etc/asterisk/pinset_1');open (PIN2,'/etc/asterisk/pinset_2');open (PIN3,'/etc/asterisk/pinset_3');$|=1;$AGI = new Asterisk::AGI;
exit if $#ARGV <0;chomp($dbid=$ARGV[0]);#$AGI->exec("NoOp","$dbid")  ;%input = $AGI->ReadParse();$try = '0';$filename = 'agent-pass';while($try < 3) {  $pin = $AGI->get_data($filename,"2000");
  if (!defined $pin) {$try++;$filename = 'auth-incorrect';next;}  if (length($pin)!=4) {$try++;$filename = 'auth-incorrect';next;}  $status=check_pin($pin);   
   if ($status =='1'){   $AGI->exec("NoOp","$pin");   $AGI->stream_file('auth-thankyou');   #$AGI->exec('Playback','auth-thankyou','noanswer');
   $AGI->exec('SetAccount',"$pin");   exit (0);   }   else {  $try++;  $filename = 'auth-incorrect';  #$AGI->exec('Playback','')  next;
  }}$AGI->exec('Playback','vm-goodbye');$AGI->exec('Wait','1');$AGI->hangup();exit (0);
 
sub check_pin{    my $pin=$_[0];    my $file='PIN'."$dbid";    seek ("$file",0,0);    while (<$file>){    if (/$pin/) {return '1';}    }    return '0';
}
 
you have to install the Asterisk::AGI module for thi script to work. i used the authentication macros from AMP (FreePBX) to pug in this agi. 
Every trunk thatrequiers authentication calls this macros. The macros is as follows..
 
 

[macro-pinsets]include => macro-pinsets-customexten => s,1,GotoIf(${ARG2} = 1?cdr,1)exten => cdr,1,AGI(auth.agi|${ARG1})
; end of [macro-pinsets]
 
i hope this will be of use to someone 
On 10/18/06, Dumpolid Exeplish <[EMAIL PROTECTED]> wrote:

I have found the problem. 
Before calls leave our network, thee user must supply a pin. this is a for of call accounting that we implemented. To do this, we had used AMP's Authenticate () function. This function actually and always answers the channel first before accepting pin entries. This was why there is always an answered flag on the channel. and since the channel is answered as soon as the call is made, there is no difference between the duration and the billsec. Now my problem is how do i implement an authentication AGI that uses DTMF ? i would be posting this question in another thread 

 
Thanks for your help 

On 10/17/06, Dumpolid Exeplish <[EMAIL PROTECTED]
> wrote: 

this Cdr Record if from the Primary PBX
 
'2006-10-17 07:11:37', 'Admin', 'XXX, 'aa', 'from-internal', 'IAX2/[EMAIL PROTECTED]' 
, 'Zap/1-1', 'ResetCDR', 'w', 10, 0, 'BUSY', 3, '', '', ''
 
 
 
this is the CDR record from the secondsry for the same call
 
 
'2006-10-17 13:31:57', '"Admin" ', 'X', 'aa', 'from-internal', 'SIP/401-8f0c', 'IAX2/TRUNK1-2', 'Dial', 'IAX2/TRUNK1/aaa|120', 15, 15, 'ANSWERED', 3, '4147', '', ''
 
in this setup, the caller dropped the call after allowing it to ring for 15 seconds

 
 
 
 
 
On 10/17/06, Dumpolid Exeplish <[EMAIL PROTECTED] 
> wrote: 

Well I am using APM on the two boxes i have modified the srripts extensievely and i am sure that there is no Awnser befor a dial when Dialing through the PBX trunks

 
 
On 10/17/06, Steve Davies <[EMAIL PROTECTED] 
> wrote: 
On 10/17/06, Dumpolid Exeplish <
 [EMAIL PROTECTED]> wrote:> Hello,> i have call time irregularites in my asterisk CDR. I a currently using a> mysqly backent to save CDR records and use this to generate bills at the end> of each month. However, my users are complaining that they gety charged for 
> even uncompleted calls ( i.e. calls they make whaich have already be setup> but canclled). i have noticed that only 'AWNSERED' and 'Busy' show up in my> call disposition colume. I have also noticed that both the call duration and 
> the billsec are always equal. here is my setup below>>   <> () <=> <> () 
>> Clients are connected to the Secondary PBX. this pbx handles registration of> all clents. The billing irregularities happen on the Secondary PBX. When a> call is maked from the Secondary and it is routed across the trunks, call 
> disposition always registeres 'AWNSERED', unless the Primary PBX sends back> a busy signal. the duration and billsecs are always equla. this means that> the user gets billed for ring time, and calls disconnected from the 
> Secondary PBX>Could you provide a snippet of the dialplan used on each of theprimary and secondary boxes to compl

Re: [asterisk-users] random one way audio and noise betweenSIP phoneson same LAN

[Giorgio Incantalupo]

Hi Scott,
so it seems that are polycom phones not working well...
have you tried with other IP phones or only with polycom?


Giorgio Incantalupo



Scott Scecina wrote:

Giorgio,

I'll answer in reverse order:

I've not had reports of "noise" from my users.  However, when I went down to
get the s/w version from the phone that seems to be acting up the most, the
user reported that earlier they were actually on a call that was ok then
spontaneously dropped the audio. Per my instructions (based on another
similar report I read on Digium's site), my user hit a digit on the phone
which brought back the caller's audio.  I've also had them attempt to put the
call on hold, and then resume, but that did not bring the audio back.

As far as the S/W versions:

One of the phones that acts up (and they all should match):

Polycom 501
BootRom: 3.1.3.0131
BootBlock: 2.5.0
SIP: 1.6.6.0036

My phone, on which I've never experienced the problem:

Polycom 601
BootRom: 3.1.3.0131
BootBlock: 2.6.0
SIP: 1.6.6.0036

- Scott


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giorgio
Incantalupo
Sent: Wednesday, October 18, 2006 11:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] random one way audio and noise betweenSIP
phoneson same LAN

Hi Scott,
seems that we have the same problem...I have canreinvite=no and polycom 
phones.
I do not have cisco switch and qualify=yes but I think that is not the 
problem.


I've got 2 questions:
1) my polycom firmware is:
sip.ver: 1.6.5.0043
bootrom.ver: 2_6_2

what are yours?
2) have you got one way calls only or noise on sip calls conversations too?

TIA


Giorgio Incantalupo

P.S.: for configuration/monitoring apps  I'm still on it...I hope to 
find useful tools asap. In case, I'll let you know.



Scott Scecina wrote:
  

I'm having the same "random" problem.

I have "canreinvite=no" on all extensions.  I have "qualify => yes" on all
non-NAT extensions. I do have several NAT extensions, but I've not had
reports of problems from those.  95% of my extensions (all polycom 501/601)
are on a brand-new network comprised of 2 48-port Cisco 3560 1GB switches.

In all cases, the called party cannot hear the calling party.  The calling
party has the "still ringing" icon on their phone, but can hear the called
party talking.  I've got call monitoring turned on, and asterisk is


recording
  
both sides of the conversation.   

The problem occurs on SIP->SIP and Zap->SIP calls. 


I've tried enabling sip debug on a particular extension that seemed to be
experiencing the problem more than others.  However the problem did not


occur
  

when the debugging was on.

Sip debug generates so much noise I've been hesitant to turn it on
system-wide.  Is there a way I can turn on sip debug and have all that
logging go to a specific file (and not in the asterisk console)?

Also, are there any other configuration/logging tricks I can try?

Thank you,

Scott Scecina





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[asterisk-users] SIP users with Database

[Maurizio Pederneschi]

Hi,
 
I'm testing Asterisk with MySql and I would want to 
insert sip users in a table "sip_users". After I modified extconfig.conf with 
"sipusers => odbc,asterisk" and I create the table sipusers, which changes 
must I make to sip.conf?
 
Thank's
Maury
 
P.S.: C'è qualche utente italiano nella mailing 
list?
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[asterisk-users] accountcode and amaflags?

[Benjamin Jacob]

Hello ppl,
Can someone explain to me the meaning and use of the variables 
accountcode and amaflags in sip.conf,etc.
Googled, voip-infoed, wikied, etc for it. Couldnt get much of it. I 
know, they are billing related, but not much beyond that.


Any ideas?

cheerz
- Ben.
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[asterisk-users] Cisco 7970 - versionStamp

[Tomislav Parčina]

If I put versionStamp in cnf.xml file, how do I check it on the phone?



--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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[asterisk-users] Re: Digium on Dell PowerEdge 1850

[Tomislav Parčina]

In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
> We're running 2 TE412P's in a Dell 1850 just fine, been running like
> this for well around 6 months to a year now without any problems.
> They're not exactly 212P's but I imagine it won't be much different.
> 
> On Wed, 2006-10-18 at 10:54 +0200, Tomislav ParÄŤina wrote:
> > Does anybody have Digium TE212P interface card on Dell PowerEdge 1850? I'm 
> > planning to install * on that configuration so I'm looking for any 
> > positive/negative experience.

Hi Aaron!

Thank you for your mail.


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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[asterisk-users] Got reject for frame XX, retransmitting frame XX now, updating n_r!

[Asterisk]

Hi all,

What does 'Got reject for frame...' message really mean, what could be
causing it, and how should one start troubleshooting it?

Thanks in advance,
Alex

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[asterisk-users] Which is the best ?

[Michel]

Hi,

As I am a newbie, I am going to ask a newbie question ;-)

I saw Digium has TE212P (with DSP) and TE210P cards  (no DSP), and 
Sangoma has A102 T1/E1 AFT card (no DSP)...


What is the best choice  of T1/E1 card (2ports) for an installation on 
Centos 4 (or simply what is the best choice of T1/E1 card ?)

? Digium  or Sangoma cards?

We would to build up a between small and medium system (and hope 
that it will grow to a large one ;-) :-P !)



Thanks you for your help



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Re: [asterisk-users] echotraining=yes in misdn.conf is invalid or out of range.

[Giorgio Incantalupo]

Hi Jarkko,
I had the same problem..It worked with an old version of misdn-install 
(taken from beronet site) but not with actual mqueue-misdn-install. I 
tried to put it in every misdn.conf section I have without success. The 
updated beronet install manual doesn't mention that parameter anymore 
so  I removed it from  misdn.conf.



Giorgio Incantalupo



Jarkko Nevala wrote:

Hi.

I'm having problems with chan_mISDN configuration. Line 
"echotraining=yes" causes warning, when Asterisk is parsing misdn.conf 
and I'm confused why the PBX doesn't accept the setting. No matter 
which section I try to offer it, it is always invalid or out of range. 
The setting itself is supposed to be valid, it is in the sample 
configuration file of chan_mISDN 0.3.1.


When I list the configuration of the ISDN-ports with "misdn show 
config", I can find values for echocancel and echocacelwhenbridged, 
but no mention about echotraining.


I'm running Asterisk 1.2.10 on OpenSuse 10.1 with 
chan_misdn-0.3.1-rc23. The hardware platform is HP server with Intel 
XEON processor and three hfcpci BRI -cards in TE -mode.


-

Here's the message I get:

 == Parsing '/etc/asterisk/misdn.conf': Found
Oct 19 00:33:54 WARNING[4443]: misdn_config.c:660 _build_port_config: 
misdn.conf: "echotraining=yes" (section: default) invalid or out of 
range. Please edit your misdn.conf and then do a "misdn reload".


-

And here is my misdn.conf:

[general]
debug=1
bridging=no
tracefile=/var/log/asterisk/misdn.trace

[default]
echocancel=yes
echotraining=yes
hold_allowed=yes
screen=-1
presentation=-1
senddtmf=yes

[isdn_call]
ports=1,2,3
context=isdn_in
msns=*

-

Have you got any idea what is causing this and how I could get the 
echo training working?


Thank you for your help.

Jarkko
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[asterisk-users] wrong outgoing caller id with PRI lines: maybe usecallingpres involved?

[Giorgio Incantalupo]

Hi,
I have a sangoma PRI card on an Asterisk PBX. I have problem with 
outgoing caller ID: when I make an outbound call, the called party gets 
x1 instead of x240 where x is the my company prefix and 240 
is the phone extensions I call from.
I read something about usecallingpres on wiki but nothing is told about 
default and possible values. Is this parameter the real cause of my 
wrong caller id?


TIA

Giorgio Incantalupo
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Re: [asterisk-users] wrong outgoing caller id with PRI lines: maybe usecallingpres involved?

[Doug Lytle]

Giorgio Incantalupo wrote:

Hi,
I have a sangoma PRI card on an Asterisk PBX. I have problem with 
outgoing caller ID: when I make an outbound call, the called party 
gets x1 instead of x240 where x is the my company prefix 
and 240 is the phone extensions I call from.
I read something about usecallingpres on wiki but nothing is told 
about default and possible values. Is this parameter the real cause of 
my wrong caller id?


How are you setting the caller id before dialing?  Show us the section 
of your dial plan that handles this.


Doug





-- Ben Franklin quote: "Those who would give up Essential Liberty to 
purchase a little Temporary Safety, deserve neither Liberty nor Safety."


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[asterisk-users] Zaptel not detecting Tormenta2 PRI Interface card

[Zeeshan Zakaria]

 
Hi,
After upgrading to new version of Asterisk and Zaptel, my Tormenta2 card has stopped working. Zaptel doesn't detect it. What should I do to make it work again. It is enabled in /etc/sysconfig/zaptel. Where else I have to enable it so zaptel can detect it and make it work. 
zaptel.conf and zapata.conf are the same as I had them before. I tried googling but couldn't find any help.-- Zeeshan A Zakaria 
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[asterisk-users] say Asterisk to answer

[Gregory Duchatelet]

Hi list,

 

I have 2 softphones, 1 Idefisk (IAX), 1 Xlite (SIP)
registered to Asterisk. One call the other-one, is it possible to order
Asterisk to force answering the call ? i.e. Xlite call Idefisk, Idefisk is
ringing, I send a command to Asterisk which force answer, so Idefisk answer the
call without clicking on “Accept” button.

 

Greg






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[asterisk-users] RE gotoiftime and Macro question

[asterisk]

Thank you all very much: it solved my problem.

exten => 554,3,GotoIfTime(08:30-14:30|mon-wed|*|*?novm,567,1)
...
[novm]
exten => _X.,1,Macro(exten-vm,novm,${EXTEN})
...

The remaining dialplan is not "my" specific,
is the standard dial plan provided by FreePbx, which I integrate in
_custom.conf file

thanks,

Andrea

Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla.

Visitate il sito http://www.frameweb.it

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Re: [asterisk-users] say Asterisk to answer

[Zoa]

It's not possible.
The idefisk however has a button to auto answer.

Zoa

Gregory Duchatelet wrote:


Hi list,

I have 2 softphones, 1 Idefisk (IAX), 1 Xlite (SIP) registered to 
Asterisk. One call the other-one, is it possible to order Asterisk to 
force answering the call ? i.e. Xlite call Idefisk, Idefisk is 
ringing, I send a command to Asterisk which force answer, so Idefisk 
answer the call without clicking on “Accept” button.


Greg



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Re: [asterisk-users] SIP users with Database

[Giovanni Miano]

Hi Maurizio,http://www.voip-info.org/wiki/view/Asterisk+RealTime+SipOften you can find what you seek just spending a minute with google
Cheers,Giovanni (Italiano)2006/10/19, Maurizio Pederneschi <[EMAIL PROTECTED]>:







Hi,
 
I'm testing Asterisk with MySql and I would want to 
insert sip users in a table "sip_users". After I modified extconfig.conf with 
"sipusers => odbc,asterisk" and I create the table sipusers, which changes 
must I make to sip.conf?
 
Thank's
Maury
 
P.S.: C'è qualche utente italiano nella mailing 
list?

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-- Giovanni Miano
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Re: [asterisk-users] say Asterisk to answer

[Giovanni Miano]

Hi Greg,Idefisk support Auto-answer only in a biz versionI suppose you got free version..You will find more details http://www.asteriskguru.com/idefisk/free/
Cheers,Giovanni2006/10/19, Gregory Duchatelet <[EMAIL PROTECTED]>:













Hi list,

 

I have 2 softphones, 1 Idefisk (IAX), 1 Xlite (SIP)
registered to Asterisk. One call the other-one, is it possible to order
Asterisk to force answering the call ? i.e. Xlite call Idefisk, Idefisk is
ringing, I send a command to Asterisk which force answer, so Idefisk answer the
call without clicking on "Accept" button.

 

Greg







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Re: [asterisk-users] Server power indication

[Erik]

I've narrowed it down to 2 configs,

It wil either be a dual:
Intel xeon dual core 3.73Ghz 1066 2x2MB cache

or a dual:
AMD Opteron DUAL CORE 285 (2.6GHz 32/64bit)

So effectively there will be 4 real CPU cores to handle processes/transcoding.
* Are there any numbers on how many (SIP-SIP) Alaw to G729 transcoding calls 
this setup will be able to handle?
* Which of these two would be the better?
* Would this configuration be enough to serve a TE412P to 120 channels G729 ?

Kind regards,
Erik


Erik wrote:
> Hello list,
> 
> I'm currently looking into building a new Asterisk server, due to some codec 
> problems i've got to transcode most of my channels between
> Alaw -- G729. Is there any indication on how many channels you would be able 
> to transcode on a certain platform?
> 
> I'm looking into  dual Xeon or dual Opteron configurations, which of these 
> platforms would perform better?
> And how much power would be needed to transcode 120 Channels PRI to G729 (for 
> example Digium TE412P)?
> 
> Erik
> ___





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Re: [asterisk-users] Electric usage of a tdm400p

[Bob Chiodini]

Erick,

It looks like the 2.5" laptop drive requires 5 watts to spin up. Adding
that to the 15 watts for the Digium card, leaves about 40 watts
available for the MB.  It's unlikely that the system will be producing
ring voltages when the drive is spinning up.  It depends on how
conservative you may be with real-time power management, e.g. spinning
drives down when not in use, etc.

I did not easily find too many ITX MB power requirements, but the one I
did find, required 45 watts (peak).

In the worst case, ringing 5 RENs on each Digium port and spinning up
the disk, you would be overtaxing the power supply.  I doubt you will
have 5 RENs on each port and all ringing, but you could.

In ages past, hard drives were the most vulnerable to poor power
regulation, but that may have changed.  With the higher cost of 2.5"
drives, I would not take any chances.  Beefing up the power supply would
also eliminate the need for manually managing power should you need a
CDROM or more power hungry drive in the future.  It's also one less
concern when troubleshooting the system.  

As Moj has pointed out, problems can occur when working close to the
edge.  I, too, have experienced similar problems when power was limited
and have had to, temporarily, resort to a "bigger" power supply to get a
system installed.  Then fell back to a "smaller" one in operation.

Good luck.

Bob...

On Wed, 2006-10-18 at 08:49 -0800, Mojo with Horan & Company, LLC wrote:
> I set up a similar system on an VIA Epia 5000, and I had issues when I 
> included the CDROM in the mix.  I had to use another ATX power supply to 
> complete the install, but then once I removed the CDROM drive I had no 
> power issues.
> 
> I presume you could install the OS with the CDROM drive installed and 
> the molex power connector REMOVED from the TDM card, then when the OS 
> was installed and you had network connectivity, power down, remove the 
> CDROM, add the power supply for the TDM card, then install zaptel etc.
> 
> Or just try it and tell us what happens, low power won't break it in my 
> experience.  Your cdrom drive might have a lower power consumption than 
> mine.
> 
> Moj
> 
> Erick Perez wrote:
> > Well Im planning to use a mini-itx, a laptop hdd and a 4fxs digium card.
> > the mini-itx comes with a 60W DC to DC adapter (80W peak).
> > So I need power to manage the hdd, motherboard,the tdm card.
> > A disk cable can be made available, but is not present as a factory default.
> > 
> > So My real concern is power.
> > 
> > 
> > On 10/18/06, Bob Chiodini <[EMAIL PROTECTED]> wrote:
> >> On Tue, 2006-10-17 at 11:59 -0500, Erick Perez wrote:
> >>> Hi people,
> >>> When you use a TDM400p with 4FXS i know i need to connect a 12V
> >>> connector to power the FXS lines.
> >>> Im not good at electric stuff so I ask...If I have a 60W DC to DC
> >>> adapter (80W peak) then, how much power will the TDM 400P consume? can
> >>> it be powered?
> >>>
> >>>
> >> Erick,
> >>
> >> Per http://en.wikipedia.org/wiki/Ring_(telephone) in the US the ring
> >> voltage is around 90VAC (20 Hz) with a current of 30 milliamperes (REN
> >> ~5).  This translates to 2.7 watts.  Assuming a DC/DC converter
> >> efficiency of 38% (probably low), you would need about 3.7 watts, per
> >> FXS module.  About 15 watts, total.
> >>
> >> What is the TDM card installed in and is a disk drive cable available?
> >>
> >> Bob...
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> > 
> > 
> 
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[asterisk-users] Access Denied on a Windows share

[Paul Gaffney]

>>> 

Message: 12

 

Date: Tue, 17 Oct 2006
18:07:04 -0700 (PDT)

 

From: sdgesa gaeharth
<[EMAIL PROTECTED]>

 

Subject: Re:
[asterisk-users] Extremely choppy sound on some of

 

 
ourPOTSnetwork    calls; goes away with mute

 

To: asterisk-users@lists.digium.com

 

Message-ID:
<[EMAIL PROTECTED]>

 

Content-Type: text/plain;
charset="iso-8859-1"

 

 

 

None of these steps have
made a difference. Any other suggestions? Here is

my original post:

 

 

 

 

Can anyone help me to figure
out why I can not write to a public share? I

was able to join the domain
without a problem.  I can access the share from

an xp box.   I
just can not write: "Access denied".

 

  

 

  thanks

 

 

 

If the person or process that is trying to write to the
share is a member of a group that is denied access then the write will fail
with “access denied”.  

 

Check the “effective permissions” for a user on
the file and look at the server security logs which may give you some
additional information.

 

Paul Gaffney

LANStatus, LLC

 

 

 






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Re: [asterisk-users] accountcode and amaflags?

[Giovanni Miano]

amaflags : Categorization for CDR records. Choices are default, omit, billing, documentation and choices are defaul, omit, billing, documentationaccountcode :  : Users may be associated with an accountcode (billing purpose)
Cheers,Giovanni2006/10/19, Benjamin Jacob <[EMAIL PROTECTED]>:
Hello ppl,Can someone explain to me the meaning and use of the variablesaccountcode and amaflags in sip.conf,etc.Googled, voip-infoed, wikied, etc for it. Couldnt get much of it. Iknow, they are billing related, but not much beyond that.
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Re: [asterisk-users] question about CDR command

[William Piper]

In cdr_mysql.conf add "userfield=1" under the globals setting.
 
bp 
On 10/18/06, unplug <[EMAIL PROTECTED]> wrote:
I want to set some custom data in the field of userfield in table CDRas following.exten => s,19,Set(CDR(userfield)=1234)
exten => s,20,Dial(SIP/1234)However, the userfield doesn't get update after making the call.After that, I relocate the command as following.exten => s,19,Dial(SIP/1234)exten => s,20,Set(CDR(userfield)=1234)
The userfield doens't get update at all.  I don't know why the fieldcan't update after issuing the command.  Anyone can help?___--Bandwidth and Colocation provided by 
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Re: [asterisk-users] accountcode and amaflags?

[Benjamin Jacob]

Giovanni,
Appreciate your lines mate.
But, Ive already read those, all over the net.

my qs inline :

amaflags : Categorization for CDR records. Choices are default, omit, 
billing, documentation and choices are defaul, omit, billing, 
documentation


wot r these categories??wot decides these categories?



accountcode :  : Users may be associated with an accountcode 
(billing purpose)


hmm.. ive seen in quite a few places, where the pin collected is stored 
as the accountcode...  wot duz that mean?
anyway, can you give me an example of wot the association means?am a lil 
slow.. 






Cheers,
Giovanni

2006/10/19, Benjamin Jacob <[EMAIL PROTECTED] 
>:


Hello ppl,
Can someone explain to me the meaning and use of the variables
accountcode and amaflags in sip.conf,etc.
Googled, voip-infoed, wikied, etc for it. Couldnt get much of it. I
know, they are billing related, but not much beyond that.

Any ideas?

cheerz
- Ben.
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--
Giovanni Miano



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Re: [asterisk-users] identifying Eicon Diva Server V-4BRI-8M vs 4BRI-8M

[Klaus Darilion]

Hi!

lspci -nv reports:

:0a:03.0 0280: 1133:e013 (rev 01)
Subsystem: 1133:e013

Thus, I suspect I really got a 4BRI-8M V2

Also divactrl reports a 4BRI:

bbgast01:~# /usr/lib/divas/divactrl ctrl -c 1 -CardName
Diva Server 4BRI-8M 2.0 PCI

Let's test faxing :-)

thanks
klaus



Armin Schindler wrote:

On Wed, 18 Oct 2006, Klaus Darilion wrote:

Hi (Armin)!

Does someone knows how to identify the type of the card? The delivery note
says it is a V-4BRI-8M, whereas lspci reports a 4BRI-8M.

What is it really? Are there any Eicon tools to identify the card type?


As far as I know these cards are almost identical, but the PCI ID must be 
different. Maybe the pci id database doesn't have this difference...


What PCI-ID does it have?
0xE012 = 4BRI-8M
0xE013 = 4BRI-8M V2
0xE016 = Voice 4BRI-8M
0xE017 = Voice 4BRI-8M V2

There is no special tool. When you load the divas driver, it should announce 
the cards found. And the divactrl utility uses divas to get the cards info 
and can tell the correct version as well, e.g.:

  divactrl ctrl -c 1 -CardName


Armin
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[asterisk-users] Bristuff qozap drivers problem

[Steve Davies]

Hi,

For a significant time now (since about 0.2.0-rc8n) the qozap driver
has become very verbose if an ISDN line is not connected... I get the
messages below every couple of seconds in the asterisk logs.

The "flaw" in the messages is the "Alarm cleared" message - The alarm
cannot possibly be cleared because there is no physical media
connected into that port!!! (BTW - All ports are in TE mode.)

Can anyone suggest a cleanup in qozap.c that will prevent it telling
Asterisk that the channel is up unless it actually has come back up? I
do not understand the zaptel/bristuff internals well enough to be able
to find where this is occuring.

I also get a solid kernel crash with no Oops if I unload the qozap
module - Again this does not happen in the older versions of the qozap
module. I am using Kernel 2.6.10.

Many thanks for any pointers,
Steve.

Oct 19 13:22:23 WARNING[6485]: chan_zap.c:6025 handle_init_event:
Detected alarm on channel 4: Red Alarm
Oct 19 13:22:23 WARNING[6485]: chan_zap.c:1445 zt_disable_ec: Unable
to disable echo cancellation on channel 4
Oct 19 13:22:23 WARNING[6485]: chan_zap.c:6025 handle_init_event:
Detected alarm on channel 5: Red Alarm
Oct 19 13:22:23 WARNING[6485]: chan_zap.c:1445 zt_disable_ec: Unable
to disable echo cancellation on channel 5
Oct 19 13:22:23 WARNING[6485]: chan_zap.c:6025 handle_init_event:
Detected alarm on channel 7: Red Alarm
Oct 19 13:22:23 WARNING[6485]: chan_zap.c:1445 zt_disable_ec: Unable
to disable echo cancellation on channel 7
Oct 19 13:22:23 WARNING[6485]: chan_zap.c:6025 handle_init_event:
Detected alarm on channel 8: Red Alarm
Oct 19 13:22:23 WARNING[6485]: chan_zap.c:1445 zt_disable_ec: Unable
to disable echo cancellation on channel 8
Oct 19 13:22:23 WARNING[6485]: chan_zap.c:6025 handle_init_event:
Detected alarm on channel 10: No Alarm
Oct 19 13:22:23 WARNING[6485]: chan_zap.c:1445 zt_disable_ec: Unable
to disable echo cancellation on channel 10
Oct 19 13:22:23 WARNING[6485]: chan_zap.c:6025 handle_init_event:
Detected alarm on channel 11: No Alarm
Oct 19 13:22:23 WARNING[6485]: chan_zap.c:1445 zt_disable_ec: Unable
to disable echo cancellation on channel 11
Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:6020 handle_init_event: Alarm
cleared on channel 4
Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:6020 handle_init_event: Alarm
cleared on channel 5
Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:6020 handle_init_event: Alarm
cleared on channel 7
Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:6020 handle_init_event: Alarm
cleared on channel 8
Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:6020 handle_init_event: Alarm
cleared on channel 10
Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:6020 handle_init_event: Alarm
cleared on channel 11
Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:8122 pri_dchannel: PRI got
event: Alarm (4) on Primary D-channel of span 2
Oct 19 13:22:23 WARNING[6485]: chan_zap.c:2197 pri_find_dchan: No
D-channels available!  Using Primary channel 6 as D-channel anyway!
Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:8122 pri_dchannel: PRI got
event: No more alarm (5) on Primary D-channel of span 2
Oct 19 13:22:23 WARNING[6485]: chan_zap.c:2197 pri_find_dchan: No
D-channels available!  Using Primary channel 6 as D-channel anyway!
Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:8122 pri_dchannel: PRI got
event: Alarm (4) on Primary D-channel of span 3
Oct 19 13:22:23 WARNING[6485]: chan_zap.c:2197 pri_find_dchan: No
D-channels available!  Using Primary channel 9 as D-channel anyway!
Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:8122 pri_dchannel: PRI got
event: No more alarm (5) on Primary D-channel of span 3
Oct 19 13:22:23 WARNING[6485]: chan_zap.c:2197 pri_find_dchan: No
D-channels available!  Using Primary channel 9 as D-channel anyway!
Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:8122 pri_dchannel: PRI got
event: Alarm (4) on Primary D-channel of span 4
Oct 19 13:22:23 WARNING[6485]: chan_zap.c:2197 pri_find_dchan: No
D-channels available!  Using Primary channel 12 as D-channel anyway!
Oct 19 13:22:23 NOTICE[6485]: chan_zap.c:8122 pri_dchannel: PRI got
event: No more alarm (5) on Primary D-channel of span 4
Oct 19 13:22:23 WARNING[6485]: chan_zap.c:2197 pri_find_dchan: No
D-channels available!  Using Primary channel 12 as D-channel anyway!
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[asterisk-users] Multiple bridge attempts

[yusuf]

Hi,

I have an Asterisk box connected via an anaogue lines(ZAP/1-1) to a Siemens PBX.  I take calls off 
the PBX and put send it to a premicell connected via ZAP/7-1.  Calls orginate from the PBX, hit 
Asterisk, then get sent to the premicell.


Can anyone tell me why there is multiple bridge attempts?  I am used to there 
been only one.

-- Starting simple switch on 'Zap/1-1'
-- Executing Dial("Zap/1-1", "ZAP/R2/0727228489|40|L(360)") in new stack
-- Called R2/0727228489
-- Zap/7-1 answered Zap/1-1
 -- Attempting native bridge of Zap/1-1 and Zap/7-1
-- Attempting native bridge of Zap/1-1 and Zap/7-1
-- Attempting native bridge of Zap/1-1 and Zap/7-1
-- Attempting native bridge of Zap/1-1 and Zap/7-1
-- Attempting native bridge of Zap/1-1 and Zap/7-1


--
thanks,
yusuf

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[asterisk-users] Please help with these SIP errors

[yusuf]

Hi,

sometimes on my Asterisk 1.2.10 box I get these errors, there are about 50 
active SIP channels so I
dont know if calls are getting dropped or not.  Should I be worried?


2006-10-18 09:33:59 WARNING[4375]: channel.c:787 channel_find_locked: Avoided 
deadlock for
'0xb7341470', 10 retries!
-- Executing GotoIf("SIP/sipCSC-b737f9e8", "0 ? 15") in new stack
2006-10-18 09:33:59 WARNING[4375]: channel.c:787 channel_find_locked: Avoided 
deadlock for
'0xb7341470', 10 retries!



== Spawn extension (iax, 0837707300, 34) exited non-zero on 
'SIP/sipCSC-b73aba28'
2006-10-18 09:33:49 ERROR[1323]: chan_sip.c:11347 sipsock_read: We could NOT 
get the channel lock
for SIP/sipCSC-b73aba28!
2006-10-18 09:33:49 ERROR[1323]: chan_sip.c:11348 sipsock_read: SIP MESSAGE 
JUST IGNORED: ACK
2006-10-18 09:33:49 ERROR[1323]: chan_sip.c:11349 sipsock_read: BAD! BAD! BAD!
  == Spawn extension (iax, 0825905581, 24) exited non-zero on 
'SIP/sipBBG-b736f910'


--
thanks,
yusuf


--
thanks,
yusuf

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Re: [asterisk-users] Re: Digium on Dell PowerEdge 1850

[Giorgio Incantalupo]

Hi Tomislav,
I use Dell hardware for desktops only. Each time I tried to use a Dell 
pc with telephony cards I get problems. It works only with a TDM400 but 
if u plan to add something more it is a real nightmare!


Giorgio Incantalupo


Tomislav Parčina wrote:

In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...
  

We're running 2 TE412P's in a Dell 1850 just fine, been running like
this for well around 6 months to a year now without any problems.
They're not exactly 212P's but I imagine it won't be much different.

On Wed, 2006-10-18 at 10:54 +0200, Tomislav ParÄŤina wrote:


Does anybody have Digium TE212P interface card on Dell PowerEdge 1850? I'm 
planning to install * on that configuration so I'm looking for any 
positive/negative experience.
  


Hi Aaron!

Thank you for your mail.


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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Re: [asterisk-users] SIP users with Database

[Maurizio Pederneschi]

Hi Giovanni,
 
I follow step by step the document that you 
suggest. The connection between asterisk and MySql works fine, but sip users 
can't resgister. 
 
If I query my sipuser table the command 
realtime loadI have not any 
result...
 
What can I check in my configuration? Can anyone 
give me some example of the configuration files in order to work with 
odbc/mysql?
 
Thank's,
Maury
 

  - Original Message - 
  From: 
  Giovanni 
  Miano 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Thursday, October 19, 2006 12:15 
  PM
  Subject: Re: [asterisk-users] SIP users 
  with Database
  Hi Maurizio,http://www.voip-info.org/wiki/view/Asterisk+RealTime+SipOften 
  you can find what you seek just spending a minute with google 
  Cheers,Giovanni (Italiano)
  2006/10/19, Maurizio Pederneschi <[EMAIL PROTECTED]>:
  

Hi,
 
I'm testing Asterisk with MySql and I would 
want to insert sip users in a table "sip_users". After I modified 
extconfig.conf with "sipusers => odbc,asterisk" and I create the table 
sipusers, which changes must I make to sip.conf?
 
Thank's
Maury
 
P.S.: C'è qualche utente italiano nella mailing 
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-- Giovanni Miano 

  
  

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[asterisk-users] Modifying SIP Stack

[German Castro]

Hello list, I am trying to include a new message after I receive a Register in 
chan_sip, at the beginning I would like to forward the same message to a fixed 
IP address, I have seen that fileds like p->sa.sin_addr and  p->sin.sin_addr  
have to be  with the IP address, but I am not sure about how to force these 
fields to be the destination that I want (192.168.1.10)... somebody knows how 
to do it? do you know where can I fond information about the "p" structure in 
asterisk and how to modify it "manually"?

Thanks and Rgds,

German



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Re: [asterisk-users] Re: Digium on Dell PowerEdge 1850

[Bruce Reeves]

Like Aaron, our asterisk systems are on Dell servers and even some Dell optiplex systems for small offices. However we use Sangoma cards to skirt compatibility issues.On 10/19/06, 
Giorgio Incantalupo <[EMAIL PROTECTED]> wrote:
Hi Tomislav,I use Dell hardware for desktops only. Each time I tried to use a Dellpc with telephony cards I get problems. It works only with a TDM400 butif u plan to add something more it is a real nightmare!
Giorgio IncantalupoTomislav Parčina wrote:> In article <[EMAIL PROTECTED]>, 
[EMAIL PROTECTED] says...>>> We're running 2 TE412P's in a Dell 1850 just fine, been running like>> this for well around 6 months to a year now without any problems.>> They're not exactly 212P's but I imagine it won't be much different.
 On Wed, 2006-10-18 at 10:54 +0200, Tomislav ParÄŤina wrote:> Does anybody have Digium TE212P interface card on Dell PowerEdge 1850? I'm planning to install * on that configuration so I'm looking for any positive/negative experience.
> Hi Aaron!>> Thank you for your mail.>>> --> Tomislav Parčina> Lama Computers Split> Stinice 12, 21000 Split> Tel.: +385(21)270248
> Mob.: +385(91)1212148> SIP: [EMAIL PROTECTED]> e-mail: tparcina#lama.hr> http://www.lama.hr> ___
> --Bandwidth and Colocation provided by Easynews.com -->> asterisk-users mailing list> To UNSUBSCRIBE or update options visit:>
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-- BruceNortex Networks
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Re: [asterisk-users] Asterisk hangs up on incoming analog calls after a while

[Eric \"ManxPower\" Wieling]

Robert La Ferla wrote:
I have been experiencing a problem where after someone calls me from an 
analog line, the phone call is terminated after a period of time 
(anywhere from 15 seconds to 15 minutes)  The phone that I use to answer 
the call is an Aastra 9133i SIP phone.  There are several other SIP 
extensions on the network as well as a few analog extensions on a shared 
FXS line.  When a call comes in the analog line on the FXO, * dials all 
the extensions (SIP and analog.)  I have a Digium card with 1 FXO and 1 
FXS.



Do you have callprogress=yes or busydetect=yes in your 
/etc/asterisk/zapata.conf ?

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Re: [asterisk-users] IAX softphones

[Administrator TOOTAI]

Guillermo Salas M. a écrit :

On Wed, 2006-10-18 at 20:08 +0200, Francesco Peeters (Asterisk) wrote:
  

On Wed, October 18, 2006 19:03, Paul Gaffney wrote:



Hi, can anyone recommend a  good IAX phone for use with Asterisk? I'm
looking for a NAT-friendly solution and my SIP phones are good but not
dependable.

Neil

Neil,

www.asteriskguru.com   lists a few of
them.  Try "IDEFISK".

Paul Gaffney

LANStatus,LLC
  

I personally like DIAX on for Windows users. Haven't yet found an IAX
phone I like on Linux...



Kiax works great with Gnome, KDE or Xfce.
  

There is also a windows version. Very nice piece of software.
--
Daniel
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Re: [asterisk-users] Access Denied on a Windows share

[Tom Vile]

Is it an NTFS Share?On 10/19/06, Paul Gaffney <[EMAIL PROTECTED]> wrote:















>>> 


Message: 12

 

Date: Tue, 17 Oct 2006
18:07:04 -0700 (PDT)

 

From: sdgesa gaeharth
<[EMAIL PROTECTED]>

 

Subject: Re:
[asterisk-users] Extremely choppy sound on some of

 

 
ourPOTSnetwork    calls; goes away with mute

 

To: asterisk-users@lists.digium.com


 

Message-ID:
<[EMAIL PROTECTED]>

 

Content-Type: text/plain;
charset="iso-8859-1"

 

 

 

None of these steps have
made a difference. Any other suggestions? Here is

my original post:

 

 

 

 

Can anyone help me to figure
out why I can not write to a public share? I

was able to join the domain
without a problem.  I can access the share from

an xp box.   I
just can not write: "Access denied".

 

  

 

  thanks

 

 


 

If the person or process that is trying to write to the
share is a member of a group that is denied access then the write will fail
with "access denied".  

 

Check the "effective permissions" for a user on
the file and look at the server security logs which may give you some
additional information.

 

Paul Gaffney

LANStatus, LLC

 

 

 







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-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com
Phone: 518-631-2855 x205Fax: 518-631-2856
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Re: [asterisk-users] Digium Single Span Card Installation

[Tzafrir Cohen]

On Thu, Oct 19, 2006 at 10:15:32AM +0530, K Y Iyer wrote:
> Hello Again
> 
> I ran the genzaptelconf - but I cannot see any channels in asterisk CLI.
> 
> I did make and install zaptel after I put in the card.  What am I
> missing?

zaptel.conf is now configured (hopefully with the correct signalling, if
not, please let me know).

genzaptelconf does not generate a complete /etc/asterisk/zapata.conf
file. Rather, it generates /etc/asterisk/zapata-channels.conf which is a
zapata.conf snippet that could be #include-d at the end of zapata.conf:

echo '#include zapata-channels.conf' >>/etc/asterisk/zapata.conf

Then you need to restart asterisk (or use 'zap restart' , if availble)

> 
> Thanks very much, indeed, for all your assistance
> 
> Best wishes
> 
> Iyer
> 
> 
> [03 ~]# asterisk -r
> 
> Asterisk 1.2.12.1, Copyright (C) 1999 - 2006 Digium, Inc. and others.
> Created by Mark Spencer <[EMAIL PROTECTED]>
> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for
> details.
> This is free software, with components licensed under the GNU General
> Public
> License version 2 and other licenses; you are welcome to redistribute it
> under
> certain conditions. Type 'show license' for details.
> 
> =
> Connected to Asterisk 1.2.12.1 currently running on mysql-03 (pid =
> 23271)
> 
> 03*CLI> zap show channels
>Chan Extension  Context Language   MusicOnHold
>  pseudodefault
> 
> 03*CLI> zap show status
> Description  Alarms IRQbpviol
> CRC4
> Digium Wildcard TE110P T1/E1 Card 0  OK 0  0
> 0
> 03*CLI> 
> 
> [03 ~]# ztcfg -vvv
> 
> Zaptel Configuration
> ==
> 
> SPAN 1: ESF/B8ZS Build-out: 133-266 feet (DSX-1)
> 
> Channel map:
> 
> Channel 01: Clear channel (Default) (Slaves: 01)
> Channel 02: Clear channel (Default) (Slaves: 02)
> Channel 03: Clear channel (Default) (Slaves: 03)
> Channel 04: Clear channel (Default) (Slaves: 04)
> Channel 05: Clear channel (Default) (Slaves: 05)
> Channel 06: Clear channel (Default) (Slaves: 06)
> Channel 07: Clear channel (Default) (Slaves: 07)
> Channel 08: Clear channel (Default) (Slaves: 08)
> Channel 09: Clear channel (Default) (Slaves: 09)
> Channel 10: Clear channel (Default) (Slaves: 10)
> Channel 11: Clear channel (Default) (Slaves: 11)
> Channel 12: Clear channel (Default) (Slaves: 12)
> Channel 13: Clear channel (Default) (Slaves: 13)
> Channel 14: Clear channel (Default) (Slaves: 14)
> Channel 15: Clear channel (Default) (Slaves: 15)
> Channel 16: D-channel (Default) (Slaves: 16)
> Channel 17: Clear channel (Default) (Slaves: 17)
> Channel 18: Clear channel (Default) (Slaves: 18)
> Channel 19: Clear channel (Default) (Slaves: 19)
> Channel 20: Clear channel (Default) (Slaves: 20)
> Channel 21: Clear channel (Default) (Slaves: 21)
> Channel 22: Clear channel (Default) (Slaves: 22)
> Channel 23: Clear channel (Default) (Slaves: 23)
> Channel 24: Clear channel (Default) (Slaves: 24)
> Channel 25: Clear channel (Default) (Slaves: 25)
> Channel 26: Clear channel (Default) (Slaves: 26)
> Channel 27: Clear channel (Default) (Slaves: 27)
> Channel 28: Clear channel (Default) (Slaves: 28)
> Channel 29: Clear channel (Default) (Slaves: 29)
> Channel 30: Clear channel (Default) (Slaves: 30)
> Channel 31: Clear channel (Default) (Slaves: 31)
> 
> 31 channels configured.
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
> Cohen
> Sent: Wednesday, October 18, 2006 5:10 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Digium Single Span Card Installation
> 
> On Wed, Oct 18, 2006 at 04:28:16PM +0530, K Y Iyer wrote:
> > Hi
> > 
> > Also tried the following command and got the following lines.  Why is 
> > that /proc/zaptel/1 reports all the 31 lines whereas ztcfg reports 18 
> > channels?  What do they mean?
> 
> ztcfg reports the channels you are about to try to configure and the
> result of that configuration. It does not scan your system for channels.
> Your /etc/zaptel.conf does not include all the bchan-nnels, and thus not
> all of them are reported and not all were configured.
> 
> 
> If you want to scan your system for channels and attempt to create a
> wonrking configuration (one that at least be able to pass ztcfg and run
> asterisk with), use xpp/utils/genzaptelconf.
> 
> 
> -- 
> Tzafrir Cohen sip:[EMAIL PROTECTED]
> icq#16849755  iax:[EMAIL PROTECTED]
> +972-50-7952406  jabber:[EMAIL PROTECTED]
> [EMAIL PROTECTED] http://www.xorcom.com
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[asterisk-users] siemens hipath interoperability - PRI/Q.SIG - card recommendation

[Pavel Jezek]

Hello, if somebody using this scenario in production successfully, 
please send me info, which ISDN card for asterisk server is usefull for 
me (Digium, Sangoma)?
my crucial requirement is "caller id name" transfer/display between ISDN 
(Siemens PBX) and IP phone connected to asterisk

I'm using PRI interface and Q.SIG signaling.
thank you
PJ


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Re: [asterisk-users] Re: 1.4 on mac OSX 10.4.8

[Dustin Wenz]

I just built 1.4.0 beta 3 on OS X 10.4.8, and it went pretty  
smoothly. I didn't need to install wget.


Asterisk starts and runs with 0% CPU. The CLI also works, but hangs  
if I try to tab-complete commands. However, that might be because I  
don't have any working config files and/or have some legacy modules  
installed from 1.2.


Now if only I could get app_conference to build, Asterisk would do  
everything I need.


- .Dustin


On Oct 18, 2006, at 3:20 AM, Martin Joseph wrote:


On 2006-10-17 14:19:00 -0700, Daniel Salama <[EMAIL PROTECTED]> said:

You can get wget for OSX from DarwinPorts (http:// 
wget.darwinports.com/)


Ok,  I bit the bullet and build wget.

This allows me to build 1.4 branch, which does the same thing as  
1.40b2.


It starts up, consumes as much CPU as is available, and is not  
responsive to CLI commands or registrations. It's a dead duck.


Anybody out there trying this stuff on OSX?

Marty


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[Asterisk-Users] rxfax problem

[Jim McIver]

Did you ever get an answer to this problem
?

 

I too am seeing this and it’s driving me mad !!!

 

Jim






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[asterisk-users] Occasional one-way audio - Sangoma A101

[Mike Clark]

We are having an occasional one w-way audio problem that occurs about 
every 25 - 30 calls on a system configured as follows:


Asterisk 1.2.12.1
Sangoma A101 w/wanpipe beta9
Polycom 500s w 1.5.3

This happens only on inbound calls from the PRI. The external caller can 
hear our customer answer and say hello, however, our customer cannot 
here their caller. Typically, the caller calls right back and all is 
fine. There is no discernable pattern as I can tell. Anyone have, and 
hopefully fix, a similar issue?


Thanks,

Mike Clark
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Re: [asterisk-users] Re: 1.4 on mac OSX 10.4.8

[Todd- Asterisk]

I'm a Certified Apple Sys Admin - lots of experience with Macs and  
Mac servers.  However, when  setting up an asterisk server, I'm still  
thinking a Dell box with linux is the best direction - to get the  
full reliability and full support of this group. Am I mistaken?  Or  
is using a Mac box just as convenient and reliable?  Or is  
traditional linux 'strongly' recommended for asterisk?  I'm looking  
at a solely IP based system - no digium cards

 thanks
   Todd
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RE: [asterisk-users] Occasional one-way audio - Sangoma A101

[Scott Scecina]

Hi Mike,

Sounds like you're having about the same problem Giorgio and I are having.
I'd be really surprised if you don't start having the same problem from
SIP-SIP calls to.  I also have a Sangoma card, and originally thought it was
only on calls coming from a PRI.  But as time has moved forward, the issue
really appears to be between the Polycoms and Asterisk.

The next time it happens, try hitting a digit (like 5) on the polycom and see
if the audio becomes available.

BTW - our other discussion on this is called "random one way audio and noise
between SIP phoneson same LAN"

- Scott

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Clark
Sent: Thursday, October 19, 2006 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Occasional one-way audio - Sangoma A101

We are having an occasional one w-way audio problem that occurs about 
every 25 - 30 calls on a system configured as follows:

Asterisk 1.2.12.1
Sangoma A101 w/wanpipe beta9
Polycom 500s w 1.5.3

This happens only on inbound calls from the PRI. The external caller can 
hear our customer answer and say hello, however, our customer cannot 
here their caller. Typically, the caller calls right back and all is 
fine. There is no discernable pattern as I can tell. Anyone have, and 
hopefully fix, a similar issue?

Thanks,

Mike Clark
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[asterisk-users] How do I configure Asterisk if I need to run Mysql server on second Linux

[Tielin Xu]

Hi List:

Please someone help me to point out where I can get the idea to
configure
Asterisk for mysql server running on different Linux.

Many thanks,

Tielin
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Re: [asterisk-users] Occasional one-way audio - Sangoma A101

[Jerry Jones]

We use almost all Polycoms, several hundred

had one way audio with 1.6.4 or 5, forget which

1.66 and 2.01 seem to be ok

We did have a few phones (2-3) that had random one way for a long  
time, replaced everything feeding them and it still happend. A month  
ago I replaced the phones and have not had a complaint since then.



On Oct 19, 2006, at 11:32 AM, Scott Scecina wrote:


Hi Mike,

Sounds like you're having about the same problem Giorgio and I are  
having.
I'd be really surprised if you don't start having the same problem  
from
SIP-SIP calls to.  I also have a Sangoma card, and originally  
thought it was
only on calls coming from a PRI.  But as time has moved forward,  
the issue

really appears to be between the Polycoms and Asterisk.

The next time it happens, try hitting a digit (like 5) on the  
polycom and see

if the audio becomes available.

BTW - our other discussion on this is called "random one way audio  
and noise

between SIP phoneson same LAN"

- Scott

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

Sent: Thursday, October 19, 2006 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Occasional one-way audio - Sangoma A101

We are having an occasional one w-way audio problem that occurs about
every 25 - 30 calls on a system configured as follows:

Asterisk 1.2.12.1
Sangoma A101 w/wanpipe beta9
Polycom 500s w 1.5.3

This happens only on inbound calls from the PRI. The external  
caller can

hear our customer answer and say hello, however, our customer cannot
here their caller. Typically, the caller calls right back and all is
fine. There is no discernable pattern as I can tell. Anyone have, and
hopefully fix, a similar issue?

Thanks,

Mike Clark
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[asterisk-users] Errors in console in every call made when using 1.4b3

[Thomas Kenyon]

When using Asterisk 1.4b3, everytime I make a call I get continuously 
(around 20 times a second) error messages like the following upon the 
call connecting (or getting to ring):


[Oct 19 17:23:22] WARNING[28682]: channel.c:767 ast_queue_frame: Unable 
to write to alert pipe on IAX2/gradwell-7, frametype/subclass 2/256 
(qlen = 96): Bad file descriptor!
[Oct 19 17:23:29] WARNING[28685]: chan_iax2.c:1834 __attempt_transmit: 
Max retries exceeded to host 84.9.159.76 on IAX2/acron-9 (type = 6, 
subclass = 1, ts=7, seqno=0)
[Oct 19 17:26:14] WARNING[31424]: channel.c:767 ast_queue_frame: Unable 
to write to alert pipe on IAX2/office-1, frametype/subclass 2/256 (qlen 
= 21): Bad file descriptor!


This did not occur with 1.4b2, Is anyone else experiencing this?

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Re: [asterisk-users] Zaptel not detecting Tormenta2 PRI Interface card

[Tzafrir Cohen]

On Thu, Oct 19, 2006 at 05:38:46AM -0400, Zeeshan Zakaria wrote:
> Hi,
> After upgrading to new version of Asterisk and Zaptel, my Tormenta2 card has
> stopped working. Zaptel doesn't detect it. What should I do to make it work
> again. It is enabled in /etc/sysconfig/zaptel. Where else I have to enable
> it so zaptel can detect it and make it work. zaptel.conf and zapata.conf are
> the same as I had them before. I tried googling but couldn't find any help.

Is the module loaded?

lsmod | grep ^zaptel

Has it detected a span?

cat /proc/zaptel/*

What do you have on /etc/zaptel.conf and on
/etc/asterisk/zapata-channels.conf ?

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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Re: [asterisk-users] Modifying SIP Stack

[burke]

I don't have access to the sip code right now, but from past network code
I've writen you could try this:

inet_aton("192.168.1.10", &p->sin.sin_addr);

If they are just wrapping the "struct sockaddr_in" as "sin" in "p". Worth
a try...

Ryan


> Hello list, I am trying to include a new message after I receive a
> Register in chan_sip, at the beginning I would like to forward the same
> message to a fixed IP address, I have seen that fileds like p->sa.sin_addr
> and  p->sin.sin_addr  have to be  with the IP address, but I am not sure
> about how to force these fields to be the destination that I want
> (192.168.1.10)... somebody knows how to do it? do you know where can I
> fond information about the "p" structure in asterisk and how to modify it
> "manually"?
>
> Thanks and Rgds,
>
> German
>
>
>
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Re: [asterisk-users] 3way calling / codec problem

[Mr. Jones]

Right -

I get the error on the console - I just can't tell how many
"transcodes" are occuring at any given point in time...

On 10/18/06, Thomas Kenyon <[EMAIL PROTECTED]> wrote:

Mr. Jones wrote:
> Is there some way I can tell?
>
> On 10/16/06, Thomas Kenyon <[EMAIL PROTECTED]> wrote:
>> Mr. Jones wrote:
>> > I'm having problems with conference calls (3-way) when I have my codec
>> > forced to g729 in sip.conf.
>> >
>> > I'm using Grandstream 2000s.
>> >
>> > If enable both g711 and g729 then 3 way calling and transfers work.
>> >
>> > I'm not sure why this would matter?
>> >
>> > Here's the error:
>> >
>> > Oct 13 13:54:45 NOTICE[31184] chan_sip.c: No compatible codecs!
>> >
>> > Any help is greatly appreciated!
>>
>> Are you out of licences? From memory when in a console each channel
>> needs to be able to be transcoded to SLIN. (where it is mixed and
>> transcoded back again).

I meant conference (not console).

You can show g729 or have a console open with verbosity set (probably to
3) and it should tell you on the console output (usually several times).

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Re: [asterisk-users] say Asterisk to answer

[Henry.L.Coleman]

The latest X-lite version has autoanswer button on the front.. marked "AA"

Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada


> Hi Greg,
>
> Idefisk support Auto-answer only in a biz version
> I suppose you got free version..
>
> You will find more details http://www.asteriskguru.com/idefisk/free/
>
> Cheers,
> Giovanni
>
> 2006/10/19, Gregory Duchatelet <[EMAIL PROTECTED]>:
>>
>>  Hi list,
>>
>>
>>
>> I have 2 softphones, 1 Idefisk (IAX), 1 Xlite (SIP) registered to
>> Asterisk. One call the other-one, is it possible to order Asterisk to
>> force
>> answering the call ? i.e. Xlite call Idefisk, Idefisk is ringing, I send
>> a
>> command to Asterisk which force answer, so Idefisk answer the call
>> without
>> clicking on "Accept" button.
>>
>>
>>
>> Greg
>>
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>>
>>
>
>
> --
> Giovanni Miano
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Re: [asterisk-users] Re: 1.4 on mac OSX 10.4.8

[Dustin Wenz]

Our company's PBX is running Asterisk 1.2 under OS X Server and it's  
been pretty reliable for the 50+ extensions that we have. The system  
has an uptime of 245 days, and no one has ever reported dropped calls  
or any other disturbing behavior. The reason we're using OS X instead  
of Linux is that it's much easier to administer in our environment.  
We have a data center full of xserves, and it would be a pain to have  
one oddball Linux box worry about. At the very least, if you started  
off with a xeon-based xserve and OS X didn't work out, you could  
always just install Linux on it.


The only glaring hole in our implementation is that we haven't found  
a good conferencing solution (no timing source = no meetme). Right  
now, we are paying a fee for a third-party service to handle that for  
us. It would be nice to get an Asterisk module working for this, or  
possibly some sort of standalone SIP conferencing server.


There are some other issues that are not a big problem for us; last I  
checked, realtime was not working for OS X ...and you're out of luck  
if you want to buy Digium hardware since they haven't developed any  
OS X drivers.


- .Dustin

On Oct 19, 2006, at 11:30 AM, Todd- Asterisk wrote:

I'm a Certified Apple Sys Admin - lots of experience with Macs and  
Mac servers.  However, when  setting up an asterisk server, I'm  
still thinking a Dell box with linux is the best direction - to get  
the full reliability and full support of this group. Am I  
mistaken?  Or is using a Mac box just as convenient and reliable?   
Or is traditional linux 'strongly' recommended for asterisk?  I'm  
looking at a solely IP based system - no digium cards

 thanks
   Todd
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[asterisk-users] plainvoip - down ???

[Joseph]

Is plainvoip down?
I've tried to contact them via email and their 800-956-3285; nobody is
answering or replying to emails

-- 
#Joseph
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Re: [asterisk-users] plainvoip - down ???

[J. Oquendo]

Joseph wrote:

Is plainvoip down?
I've tried to contact them via email and their 800-956-3285; nobody is
answering or replying to emails

  

I can get there just fine. Your routes might be toasted

[EMAIL PROTECTED] ~]# ping -c 10 plainvoip.com
PING plainvoip.com (66.199.240.2) 56(84) bytes of data.
...
--- plainvoip.com ping statistics ---
10 packets transmitted, 10 received, 0% packet loss, time 9013ms
rtt min/avg/max/mdev = 75.531/78.550/80.349/1.418 ms, pipe 2

Depending on your location thought, there are issues with GBLX possibly 
due to a fiber cut either in VA or DC.


--

J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743
sil . infiltrated @ net http://www.infiltrated.net 


The happiness of society is the end of government.
John Adams



smime.p7s
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Re: [asterisk-users] How do I configure Asterisk if I need to run Mysql server on second Linux

[Steve Edwards]

If you are using MySQL for storing CDR's, this is what I use (slightly 
modified, of course):


cat /etc/asterisk/cdr_mysql.conf
;
[global]
dbname  = 
hostname= 
password= 
port= 3306
sock= /tmp/mysql.sock
table   = cdrs
user= 
userfield   = 1

On Thu, 19 Oct 2006, Tielin Xu wrote:


Hi List:

Please someone help me to point out where I can get the idea to
configure
Asterisk for mysql server running on different Linux.

Many thanks,

Tielin
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Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] plainvoip - down ???

[Jeremy McNamara]

Joseph wrote:

Is plainvoip down?
I've tried to contact them via email and their 800-956-3285; nobody is
answering or replying to emails



Since when is the asterisk-users list second level support for VoIP 
providers?



If they are down, I am sure they are well aware of it.



Jeremy McNamara
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[asterisk-users] T1 pricing in Oz

[Forum]

I’m looking at getting a T1 into a location in
Melbourne, Australia and was wondering if anyone has a good source and pricing
for this.

 

Cheers,

 

Steve

 

 






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Re: [asterisk-users] T1 pricing in Oz

[Dennis Gilmore]

On Thursday 19 October 2006 14:43, Forum wrote:
> I'm looking at getting a T1 into a location in Melbourne, Australia and was
> wondering if anyone has a good source and pricing for this.
I think your looking for a E1  Australia follows the European standard  last i 
looked.  I never did any voice stuff when i lived in Oz.  but you could try 
Telstra, Optus, etc,  or look in the phone book for a reseller.   Someone  
probably has alot better experience than me.

Dennis
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[asterisk-users] Embedded Asterisk

[Cory Andrews]

I caught a thread the other day concerning Astricon and users embedding
Asterisk on a Linksys or Netgear broadband router.  I lost track of the
email thread, if anyone is presently working with this scenario please shoot
me an email.

Thanks

Cory Andrews
++
VoIPSupply.com
PBXSelect.com
++
454 Sonwil Drive
Buffalo, NY 14225
voice direct - 716.250.3402
fax - 716.630.1548
e - [EMAIL PROTECTED]
m - 716.907.4059
aim - B2Cory

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Re: [asterisk-users] Embedded Asterisk

[Kristian Kielhofner]

Cory Andrews wrote:

I caught a thread the other day concerning Astricon and users embedding
Asterisk on a Linksys or Netgear broadband router.  I lost track of the
email thread, if anyone is presently working with this scenario please shoot
me an email.

Thanks

Cory Andrews
++
VoIPSupply.com
PBXSelect.com
++
454 Sonwil Drive
Buffalo, NY 14225
voice direct - 716.250.3402
fax - 716.630.1548
e - [EMAIL PROTECTED]
m - 716.907.4059
aim - B2Cory



Cory,

	Brian Capouch is very interested in running Asterisk on Mipsel hardware 
like the Linksys WRT, etc.  His latest favorite is the Netgear WGT634U 
(it has a USB port for extra storage and a miniPCI slot for a wifi 
radio).  It's also dirt cheap (just the way he likes it). 
Unfortunately, Netgear has discontinued this model (about a year ago - I 
think), so you better get them while they're hot!


	Brian will be helping people flash the the Netgear's with the latest 
version of the firmware + Asterisk.


	He is very dedicated to reading the list, I'm sure he'll get back to 
you too.


--
Kristian Kielhofner
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[asterisk-users] DTMF / Silence issues

[Jason Walker]

I am now running 1.4 beta3
I have an ongoing issue that it does not recognize my DTMF key press. I
will call and press as many numbers and the background message still plays.
I am also having an issue with transfers
NOTICE[30930]: chan_sip.c:13289 handle_request_invite: Unable to
create/find SIP channel for this INVITE
happens everytime

Any ideas.

I tried to go back to 1.2 and the modules would not show up.

Thanks
Jason


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Re: [asterisk-users] considering purchasing a t1 card, any recommendations?

[Dovid B]

Can I now 5th it ? All this makes me wonder 
why Digium dosent work harder. I have mainly only seen others praise Sangoma 
over Digium.

  - Original Message - 
  From: 
  Tom Vile 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Wednesday, October 18, 2006 4:22 
  PM
  Subject: Re: [asterisk-users] considering 
  purchasing a t1 card,any recommendations?
  I 4th it.
  On 10/18/06, Matthew 
  Thompson <[EMAIL PROTECTED]> 
  wrote:
  


On 17 Oct 2006, at 22:09, Richard wrote:

  I would have 
  to second the Sangoma buy.  Their tech support is second to none and 
  more then helpful. 
   
  I've never 
  had any problems with their products that wasn't my own fault. 
  
Thirded - I've just done another install with a Sangoma A102 
- the setup guides you through all the way and takes no more than 30 minutes 
(Including recompiling zaptel, which it does for you) 

[EMAIL PROTECTED] :o)


-- 
Matthew Thompson
[EMAIL PROTECTED]

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-- Tom 
  VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP 
  Telephonywww.baldwintechsolutions.com 
  Phone: 518-631-2855 x205Fax: 518-631-2856 
  
  

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Re: [asterisk-users] Findme problem

[Dovid B]

I have a backup of a working version on my server 
some where. When i find it I will post it.
 
Dovid
 

  - Original Message - 
  From: 
  Eric 
  Jacksch 
  To: asterisk-users@lists.digium.com 
  
  Sent: Wednesday, October 18, 2006 4:20 
  PM
  Subject: [asterisk-users] Findme 
  problem
  
  Greetings all,
   
  I've been working on having Asterisk put a call 
  through to two different numbers, and give the call to the first one that 
  acknowledges by pressing the 1 key.  I found an example on the wiki, but 
  I can't get it working.
   
  When I answer the call I hear the message telling 
  me to press 1 to connect, and as soon as the message is done, the call is 
  connected.  In other words, it is not waiting for me to press a 
  key.
   
  I'm sure this is a forehead slapper, but I just 
  can't see it...can anyone help?  Here's the relevant portion of the 
  dialplan,  It executes the NoOp(Waiting) and then the macro seems to 
  immediately exit and the call is connected.
   
  [default]exten => 
  _XX,1,Dial(SIP/provider/${EXTEN:4},40,M(screen))exten => 
  _XX,2,Hangup
   
  [macro-screen]exten => 
  s,1,Wait(1)exten => s,2,Set(TIMEOUT(digit)=5)exten => 
  s,3,Set(TIMEOUT(response)=10)exten => s,4,Background(press-1)exten 
  => s,5,NoOp(Waiting)
   
  exten => 1,1,NoOp(Caller 
accepted)
   
  exten => i,1,NoOp(Invalid 
  response)
  exten => 
  i,2,Set(MACRO_RESULT=CONTINUE)
  exten => t,1,NoOp(Timeout)exten => 
  t,2,Set(MACRO_RESULT=CONTINUE)
   
  [find-eric]exten => 
  s,1,Playback(pls-wait-connect-call)exten => 
  s,n,Dial(LOCAL/6135551212&LOCAL/6135551313,40,m)
   
  (I have replaced the phone numbers with bogus 
  ones).
   
  Thanks,
  Eric
  
  

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Re: [asterisk-users] Findme problem

[Dovid B]

Ooops. I read the email wrong. The macro I created 
called one number. If the person didnt accept the call or if they didnt pick up 
then it tried the second person. Let me know if you still want it.
 
Dovid
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Re: [asterisk-users] siemens hipath interoperability - PRI/Q.SIG - cardrecommendation

[wendys]

Hi,

we have tested the Digium-Cards, they work fine, but don't expect to much!
Only segmentation 1 in Ecma (it is not a digium-problem)
The Name ist displayed, but only in Hex-Code (this is due to the 
Libpri/Zaptel Drivers but I didn't fint a way to display it in *)
There is also very less documentation, on Asterisk.org (Features) there is 
non Q.Sig Support offered.

Also very less documentation through google available.
;-(

If you find some hints, i'm also interested!


Regards wendy

- Original Message - 
From: "Pavel Jezek" <[EMAIL PROTECTED]>

To: 
Sent: Thursday, October 19, 2006 5:22 PM
Subject: [asterisk-users] siemens hipath interoperability - PRI/Q.SIG - 
cardrecommendation



Hello, if somebody using this scenario in production successfully, please 
send me info, which ISDN card for asterisk server is usefull for me 
(Digium, Sangoma)?
my crucial requirement is "caller id name" transfer/display between ISDN 
(Siemens PBX) and IP phone connected to asterisk

I'm using PRI interface and Q.SIG signaling.
thank you
PJ


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Re: [asterisk-users] considering purchasing a t1 card, any recommendations?

[Conrad Wood]

On Thu, 2006-10-19 at 22:39 +0200, Dovid B wrote:
> Can I now 5th it ? All this makes me wonder why Digium dosent work
> harder. I have mainly only seen others praise Sangoma over Digium.

I strongly suspect digium is painfully aware of the problems with some 
combinations of mboards and their cards,
but given limited resources, what would you focus development efforts
on? Creating echo-cancellers and bri cards or designing a PCI interface
that works even with the most broken motherboard chipset?
Most people who have digium cards working seem to have them working
extremely well and some can't get them to work at all.
Frankly, I have seen so many motherboards that I consider outright
broken, I wouldn't blame a digium card for not working in any of those.
In fact, those motherboards tend to have problems with most cards, it
just happens to show much more when you try and do voice traffic (e.g.
isdn) rather than say IP traffic.
I'm not qualified to judge digium vs sangoma isdn facing interfaces, but
I'd be interested how that compares, if someone can shed some light on
that, I'd read it with interest.
E.g., how do digium cards perform on substandard isdn lines? How do they
handle faults on the line? How compatible (compared to sangoma) are they
with different exchange equipment?
Last but not least, might that be why digium secured some funding
recently, to increase the resources for development on these cards?

just my 2 pence worth ;)


Conrad


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Re: [asterisk-users] Occasional one-way audio - Sangoma A101

[Mike Clark]

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Mike Clark
> Sent: Thursday, October 19, 2006 12:07 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Occasional one-way audio - Sangoma A101
> 
> We are having an occasional one w-way audio problem that occurs about 
> every 25 - 30 calls on a system configured as follows:
> 
> Asterisk 1.2.12.1
> Sangoma A101 w/wanpipe beta9
> Polycom 500s w 1.5.3
> 
> This happens only on inbound calls from the PRI. The external caller can 
> hear our customer answer and say hello, however, our customer cannot 
> here their caller. Typically, the caller calls right back and all is 
> fine. There is no discernable pattern as I can tell. Anyone have, and 
> hopefully fix, a similar issue?
> 
> Thanks,
> 
> Mike Clark

Scott Scecina wrote:
> Hi Mike,
>
> Sounds like you're having about the same problem Giorgio and I are having.
> I'd be really surprised if you don't start having the same problem from
> SIP-SIP calls to.  I also have a Sangoma card, and originally thought
it was
> only on calls coming from a PRI.  But as time has moved forward, the issue
> really appears to be between the Polycoms and Asterisk.
>
> The next time it happens, try hitting a digit (like 5) on the polycom
and see
> if the audio becomes available.
>
> BTW - our other discussion on this is called "random one way audio and
noise
> between SIP phoneson same LAN"
>
> - Scott
>

So, do newer versions of Polycom firmware, like 1.6.7, help? I saw
Asterisk 1.2.13 is now available. Have you tried that. The info on the
Asterisk site says "numerous" bugs have been fixed.

Mike
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RE: [asterisk-users] T1 pricing in Oz

[Forum]

Thanks Dennis,

I am an Australian living in Canada and like yourself have done no voice
work
back home


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dennis Gilmore
Sent: Thursday, October 19, 2006 1:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T1 pricing in Oz

On Thursday 19 October 2006 14:43, Forum wrote:
> I'm looking at getting a T1 into a location in Melbourne, Australia 
> and was wondering if anyone has a good source and pricing for this.
I think your looking for a E1  Australia follows the European standard  last
i looked.  I never did any voice stuff when i lived in Oz.  but you could
try 
Telstra, Optus, etc,  or look in the phone book for a reseller.   Someone  
probably has alot better experience than me.

Dennis
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[asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now

[Vidar]

Bristuff has been updated;

http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1u.tar.gz


--
Vidar
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[asterisk-users] DTMF logging

[Scott England]

If I set the logging.conf to log DTMF it only seems to log dtmf messages
that are bridged through the * server. If the call goes into a menu the
DTMF dont get logged. Is the intended behavior? 

Scott England

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Re: [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now

[Michiel van Baak]

On 23:04, Thu 19 Oct 06, Vidar wrote:
> Bristuff has been updated;
> 
> http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1u.tar.gz

Thanks for the information.

It's a shame we need to read this here and not see it on
their website.
-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer afficionados are both called users?"

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RE: [asterisk-users] Occasional one-way audio - Sangoma A101

[Scott Scecina]

Mike,

I'm not able to download the newer polycom software(s) 'cause I'm not
"certified" in anything.

I just saw the announcement on the Asterisk updates today. I'll get them
installed tonight... 

- Scott

> Scott Scecina wrote:
> > Hi Mike,
> >
> > Sounds like you're having about the same problem Giorgio and I are
> having.
> > I'd be really surprised if you don't start having the same problem from
> > SIP-SIP calls to.  I also have a Sangoma card, and originally thought
> it was
> > only on calls coming from a PRI.  But as time has moved forward, the
> issue
> > really appears to be between the Polycoms and Asterisk.
> >
> > The next time it happens, try hitting a digit (like 5) on the polycom
> and see
> > if the audio becomes available.
> >
> > BTW - our other discussion on this is called "random one way audio and
> noise
> > between SIP phoneson same LAN"
> >
> > - Scott
> >
> 
> So, do newer versions of Polycom firmware, like 1.6.7, help? I saw
> Asterisk 1.2.13 is now available. Have you tried that. The info on the
> Asterisk site says "numerous" bugs have been fixed.
> 
> Mike
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Re: [asterisk-users] considering purchasing a t1 card, any recommendations?

[Zoa]

I think the recent Digium and Sangoma cards are quite similar. (and 
about the same price)
I didn't try sangoma so far, never had any issues with the digium cards, 
I have no clue how the digium helpdesk is, i never needed to call them.
(well not really correct i did call them once, years ago for a firmware 
problem with their first te410p revision, causing a crash once every few 
months they had the distributor send me replacement cards right away, 
before i returned the old ones, so that i could swap them without having 
to shut down the server for a week).


Configuration and installation for the cards is pretty straightforward, 
all you need to do is compile the kernel modules for your kernel.


I personally installed at least 20 digium pri cards, all on different 
hardware without problems related to the digium hardware. (sometimes i 
did have bad cables, bad pri's, oh and my embedded pc didn't provide 
enough power for FXO ports).


You will probably find more people on the list with problems with digium 
than people with problems with sangoma. This might be because a lot more 
people seem to use the digium cards with asterisk than sangoma cards 
with asterisk. (Based on the people i speak to, i'd guess 1 to 5% use 
sangoma?).


The biggest choice you need to make is if you want onboard echo 
cancellation or not, you might not need it and if you want it its going 
to cost you a lot more than without. (both for sangoma and digium 
hardware). - They both seem to use exactly the same Octasic echo 
cancellation module.


If you need on board echo cancellation but don't need 4 ports, digium is 
the only choice with their 2 port card with Octasic echo cancellation 
module.
(Afaik sangoma doesn't have such a 2 port board with on board E.C. but i 
could be wrong.)


Btw, there are more options, dialogic has compatible cards and so does 
eicon. (you will need deeper pockets though, the eicon retails at +/- 
12000 euro for a quad span i think - people who buy these for asterisk 
usually do so for hardware faxing or interconnection to different 
carriers at the same time.)


Some people prefer digium over sangoma because they sponsor the asterisk 
development that way.  I'm not one of them, i buy digium cards (or tell 
my customers to buy them) because i'm happy with their product.


Dislaimer: I know some of the people within Digium quite well, so maybe 
i get exceptional support or they ship me handpicked gold plated, 
overclocked versions of their cards (not really since i just buy them 
from a reseller).


Cheers,

Zoa.

Dovid B wrote:
Can I now 5th it ? All this makes me wonder why Digium dosent work 
harder. I have mainly only seen others praise Sangoma over Digium.


- Original Message -
*From:* Tom Vile 
*To:* Asterisk Users Mailing List - Non-Commercial Discussion

*Sent:* Wednesday, October 18, 2006 4:22 PM
*Subject:* Re: [asterisk-users] considering purchasing a t1
card,any recommendations?

I 4th it.

On 10/18/06, *Matthew Thompson* <[EMAIL PROTECTED]
> wrote:


On 17 Oct 2006, at 22:09, Richard wrote:


I would have to second the Sangoma buy.  Their tech support
is second to none and more then helpful.
 
I've never had any problems with their products that wasn't

my own fault.


Thirded - I've just done another install with a Sangoma A102 -
the setup guides you through all the way and takes no more
than 30 minutes (Including recompiling zaptel, which it does
for you)

[EMAIL PROTECTED] :o)

-- 
Matthew Thompson

[EMAIL PROTECTED]






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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] PRI Card

[Duracom Lists]

We are looking at migrating our office from a Samsung PBX to an Asterisk
PBX.  I am looking at ordering a PRI with 12 Channels for now (we currently
have 8 analog lines) and need to know what PRI card you guys would recommend
that we use.  I have seen some with Echo Cancellation and so on, but don't
know which one would be best to get.

K

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Re: [asterisk-users] Electric usage of a tdm400p

[Mojo with Horan & Company, LLC]

Further, I should have mentioned my resolution.  As Bob mentioned, 
upgrade the power supply now to give you options and peace of mind in 
the future.  I am using a picoPSU-120 in the aforementioned itx box, for 
example,


http://www.mini-box.com/s.nl/it.A/id.417/.f?category=13

and have been quite happy with it :)

Moj


Bob Chiodini wrote:

Erick,

It looks like the 2.5" laptop drive requires 5 watts to spin up. Adding
that to the 15 watts for the Digium card, leaves about 40 watts
available for the MB.  It's unlikely that the system will be producing
ring voltages when the drive is spinning up.  It depends on how
conservative you may be with real-time power management, e.g. spinning
drives down when not in use, etc.

I did not easily find too many ITX MB power requirements, but the one I
did find, required 45 watts (peak).

In the worst case, ringing 5 RENs on each Digium port and spinning up
the disk, you would be overtaxing the power supply.  I doubt you will
have 5 RENs on each port and all ringing, but you could.

In ages past, hard drives were the most vulnerable to poor power
regulation, but that may have changed.  With the higher cost of 2.5"
drives, I would not take any chances.  Beefing up the power supply would
also eliminate the need for manually managing power should you need a
CDROM or more power hungry drive in the future.  It's also one less
concern when troubleshooting the system.  


As Moj has pointed out, problems can occur when working close to the
edge.  I, too, have experienced similar problems when power was limited
and have had to, temporarily, resort to a "bigger" power supply to get a
system installed.  Then fell back to a "smaller" one in operation.

Good luck.

Bob...

On Wed, 2006-10-18 at 08:49 -0800, Mojo with Horan & Company, LLC wrote:
I set up a similar system on an VIA Epia 5000, and I had issues when I 
included the CDROM in the mix.  I had to use another ATX power supply to 
complete the install, but then once I removed the CDROM drive I had no 
power issues.


I presume you could install the OS with the CDROM drive installed and 
the molex power connector REMOVED from the TDM card, then when the OS 
was installed and you had network connectivity, power down, remove the 
CDROM, add the power supply for the TDM card, then install zaptel etc.


Or just try it and tell us what happens, low power won't break it in my 
experience.  Your cdrom drive might have a lower power consumption than 
mine.


Moj

Erick Perez wrote:

Well Im planning to use a mini-itx, a laptop hdd and a 4fxs digium card.
the mini-itx comes with a 60W DC to DC adapter (80W peak).
So I need power to manage the hdd, motherboard,the tdm card.
A disk cable can be made available, but is not present as a factory default.

So My real concern is power.


On 10/18/06, Bob Chiodini <[EMAIL PROTECTED]> wrote:

On Tue, 2006-10-17 at 11:59 -0500, Erick Perez wrote:

Hi people,
When you use a TDM400p with 4FXS i know i need to connect a 12V
connector to power the FXS lines.
Im not good at electric stuff so I ask...If I have a 60W DC to DC
adapter (80W peak) then, how much power will the TDM 400P consume? can
it be powered?



Erick,

Per http://en.wikipedia.org/wiki/Ring_(telephone) in the US the ring
voltage is around 90VAC (20 Hz) with a current of 30 milliamperes (REN
~5).  This translates to 2.7 watts.  Assuming a DC/DC converter
efficiency of 38% (probably low), you would need about 3.7 watts, per
FXS module.  About 15 watts, total.

What is the TDM card installed in and is a disk drive cable available?

Bob...
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!DSPAM:500,453763ec326532002735277!



--
Mojo <[EMAIL PROTECTED]>
Office Manager, Horan & Company, LLC
(907) 747- x112
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Re: [asterisk-users] plainvoip - down ???

[Andres]

Joseph wrote:


Is plainvoip down?
I've tried to contact them via email and their 800-956-3285; nobody is
answering or replying to emails

 


This is starting to sound like a rerun of Livevoip.  Remember that company?
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Re: [asterisk-users] Embedded Asterisk

[Michael Graves]

On Thu, 19 Oct 2006 16:10:23 -0400, Cory Andrews wrote:

>I caught a thread the other day concerning Astricon and users embedding
>Asterisk on a Linksys or Netgear broadband router.  I lost track of the
>email thread, if anyone is presently working with this scenario please shoot
>me an email.


Cory,

I've been running Astlinux on an H-P T5700 thin client. These are available, 
not dirt cheap, but I received box of them that were due to be recycled. I see 
them on Ebay for around $80-150. 
That's less than a Soekris Net4801 new in a case.

They have the advantage of some built-in flash in an IDE DOM card, then 4 USB 
ports as well as traditional mouse, keyboard, VGA ports. Also, the CPUs which 
are in the 800 MHz - 1.2 GHz 
range handle G.729 codecs better than the Net4801 or WRAP boards.

Michael Graves



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Re: [asterisk-users] considering purchasing a t1 card, any recommendations?

[John Novack]

Conrad Wood wrote:

On Thu, 2006-10-19 at 22:39 +0200, Dovid B wrote:
  

Can I now 5th it ? All this makes me wonder why Digium dosent work
harder. I have mainly only seen others praise Sangoma over Digium.



I strongly suspect digium is painfully aware of the problems with some 
combinations of mboards and their cards,
but given limited resources, what would you focus development efforts on? 
Creating echo-cancellers and bri cards or designing a PCI interface that works 
even with the most broken motherboard chipset?
  
Sangoma seems to accomplish that quite well. In fact they clearly state 
their cards will work

Digium says "try another motherboard"
That answer is an insult
Especially when the same motherboard/machine/software works with the Sangoma
Their answer really should be "buy a Sangoma"
Perhaps their efforts would be better spent as a Sangoma reseller.

Just my opinion.
Worth what you paid for it

John Novack

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Re: [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now

[Tzafrir Cohen]

On Thu, Oct 19, 2006 at 11:27:07PM +0200, Michiel van Baak wrote:
> On 23:04, Thu 19 Oct 06, Vidar wrote:
> > Bristuff has been updated;
> > 
> > http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1u.tar.gz
> 
> Thanks for the information.
> 
> It's a shame we need to read this here and not see it on
> their website.

Also note that the changelog entry for 0.3.0-PRE-1u is missing from the
CHANGES file. Nevertheless, that is a version for Asterisk 1.2.13 ,
Zaptel 1.2.10 and libpri 1.2.4 .

-- 
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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[asterisk-users] FS: Sangoma A200 10 port FXO card

[Mark Phillips]

Hi folks,

I have a Sangoma A200 10 port FXO card for sale.

US$500 secures plus shipping. 


Thanks

Mark

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Re: [asterisk-users] question about CDR command

[unplug]

Thanks!!
Just one more question.  Can I do the same "add fieldname=1" if I add
a field "fieldname" in the cdr table to perform the same action?

On 10/19/06, William Piper <[EMAIL PROTECTED]> wrote:

In cdr_mysql.conf add "userfield=1" under the globals setting.

bp


On 10/18/06, unplug <[EMAIL PROTECTED]> wrote:
>
> I want to set some custom data in the field of userfield in table CDR
> as following.
> exten => s,19,Set(CDR(userfield)=1234)
> exten => s,20,Dial(SIP/1234)
>
> However, the userfield doesn't get update after making the call.
> After that, I relocate the command as following.
>
> exten => s,19,Dial(SIP/1234)
> exten => s,20,Set(CDR(userfield)=1234)
>
> The userfield doens't get update at all.  I don't know why the field
> can't update after issuing the command.  Anyone can help?
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Re: [asterisk-users] T1 pricing in Oz

[Paul Hales]

E1 is readily available in Australia, and very easy to set up.

later,

PaulH

On Thu, 2006-10-19 at 15:05 -0500, Dennis Gilmore wrote:
> On Thursday 19 October 2006 14:43, Forum wrote:
> > I'm looking at getting a T1 into a location in Melbourne, Australia and was
> > wondering if anyone has a good source and pricing for this.
> I think your looking for a E1  Australia follows the European standard  last 
> i 
> looked.  I never did any voice stuff when i lived in Oz.  but you could try 
> Telstra, Optus, etc,  or look in the phone book for a reseller.   Someone  
> probably has alot better experience than me.
> 
> Dennis
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Re: [asterisk-users] asterisk to asterisk DID extentions

[Paul Hales]

On Fri, 2006-10-13 at 01:55 +1000, Matt wrote:
> Avi Miller wrote:
> >
> > On 04/10/2006, at 1:55 AM, Matt wrote:
> >
> >> How can I make * aware of the other ext on the remote box so the DID 
> >> caller can access them like he can with the local box?
> >
> > On each box, define the other range:
> >
> > Box A:
> >
> > exten => _9XX,1,Dial(IAX2/BoxB/${EXTEN})
> >
> > Box B:
> >
> > exten => _8XX,1,Dial(IAX2/BoxA/${EXTEN})
> >
> Thanks Avi but it didnt work.  It still says "not a valid extention" 
> from the indial :(
> 
> -Matt

Is IAX working fine between the boxes? 

Or is this a dial-in issue (8XXX,1,Dial(IAX2/BoxA/${EXTEN:5}) ???

later,

PaulH

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Re: [asterisk-users] T1 pricing in Oz

[Callum McGillivray]

Dennis, I work for a company that can provide E1's in a secure data
center in Melbourne CBD.

Contact me off list for a quote etc.

Paul Hales wrote:

  E1 is readily available in Australia, and very easy to set up.

later,

PaulH

On Thu, 2006-10-19 at 15:05 -0500, Dennis Gilmore wrote:
  
  
On Thursday 19 October 2006 14:43, Forum wrote:


  I'm looking at getting a T1 into a location in Melbourne, Australia and was
wondering if anyone has a good source and pricing for this.
  

I think your looking for a E1  Australia follows the European standard  last i 
looked.  I never did any voice stuff when i lived in Oz.  but you could try 
Telstra, Optus, etc,  or look in the phone book for a reseller.   Someone  
probably has alot better experience than me.

Dennis
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[asterisk-users] Re: 1.4 on mac OSX 10.4.8

[Martin Joseph]

On 2006-10-19 08:51:01 -0700, Dustin Wenz <[EMAIL PROTECTED]> said:

I just built 1.4.0 beta 3 on OS X 10.4.8, and it went pretty  smoothly. 
I didn't need to install wget.


Asterisk starts and runs with 0% CPU. The CLI also works, but hangs  if 
I try to tab-complete commands. However, that might be because I  don't 
have any working config files and/or have some legacy modules  
installed from 1.2.


Now if only I could get app_conference to build, Asterisk would do  
everything I need.


ok,  I don't really get that, unless the make file was changed...

When you say it runs with 0% of the CPU, you mean it's not eating the 
whole CPU right?  Mine is 0% available.


I did kill all the warned about modules...  So I am not clear why I 
have no joy with 1.4b3.  I am going to keep building 1.4 branch for a 
while and see if it resolves itself,


This was a fully working 1.2.12 box that had been a spare for me.

Oh well...  Please do let us know if you configure it and are able to 
accepts registrations and make it work?


Thanks,
Marty


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[asterisk-users] Re: 1.4 on mac OSX 10.4.8

[Martin Joseph]

On 2006-10-19 09:30:14 -0700, Todd- Asterisk 
<[EMAIL PROTECTED]> said:


I'm a Certified Apple Sys Admin - lots of experience with Macs and  Mac 
servers.  However, when  setting up an asterisk server, I'm still  
thinking a Dell box with linux is the best direction - to get the  full 
reliability and full support of this group.

The support thing is the strongest argument above.

 Am I mistaken?  Or  is using a Mac box just as convenient and reliable?
My old imac G3/400 has been chugging along for months without any 
issues. It doesn't handle a lot of calls (about 50 a day),  but it 
doesn't have any problems either.


It has about 6 extensions, some SIP and some IAX2 as well as PSTN 
gateway (wellgate 3701a).  I have some echo across the gateway (I have 
a long loop) but that resolves within second of a call begining and 
seems to be a standard issue around these parts (mac or Linux).  I also 
have 3 IAX call terminators for US 48 AND Canada, as well as one 
inbound DID (just for testing really).
  Or is  traditional linux 'strongly' recommended for asterisk?  I'm 
looking  at a solely IP based system - no digium cards
If you don't need Zaptel for hardware support, it seems to me that 
Asterisk 1.09-1.2.12 has been solid on OSX (up till 1.4).


Plus, if you buy Apple hardware new, it's firkin' intel hardware 
anyhow, so you could always boot it as a linux box if need be ;~)




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[asterisk-users] question about asterisk txFAX

[Darren]

I have installed the libtiff(3.5.7),spandsp-0.0.2pre24,app_txfax and
app_rxfax,asterisk-1.2.12.1 on the CentOS 4.2.

I set sip.conf like this:
[sip_local]
host=192.168.2.111
type=friend
dtmfmode=rfc2833
canreinvite=no
insecure=very

I set extensions.conf like this:
[test]
exten=>_9001,1,Set(CALLERID=${EXTEN})
exten=>_9001,n,dial(SIP/[EMAIL PROTECTED])
exten=>_9001,n,hangup

[fax]
exten=>_9002,n,SetVar(FAXFILE=/var/spool/asterisk/fax/test.tif)

exten=>_9002,n,SetVar(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLEEID})
exten=>_9002,n,txfax(${FAXFILE})

hen I dial 9001,it's OK,but when I dial 9002,it faild fax,I
thought that maybe I had not set the sip_local in the [fax] config
session,but how can I set that?

I search on the google net, but have no idears about this,how can I
send the fax out?

Thanks for any hints!
fangh.




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[asterisk-users] Polycom boot error

[Neider, Clint]

I am having the same issue as below.  Has this issue been  solved or does anyone know an
answer?  This error recently began and we
have multiple phones out of commission. 
PLEASE HELP!!

 

http://lists.digium.com/pipermail/asterisk-users/2006-August/162841.html

 

How did you find out about 468*??? It's sure as poop not documented in the Polycom Admin Guide anywhere. -Original Message-From: Dovid Bender [mailto:asteriskusers at dovid.net]Sent: Tuesday, August 15, 2006 11:16 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Polycom upgrade issue  I believe 468* resets the phone but dosent return it to the orig. firmware. Also try to name the files with the phones mac id and see what happens. I am doing this with 1.6.6 and its working fine. - Original Message - From: Curt   Shaffer To: 'Asterisk Users Mailing List -   Non-Commercial Discussion' Sent: Tuesday, August 15, 2006 10:07 PMSubject: [asterisk-users] Polycom upgrade issue  OK, I may have done something stupid. I was trying to upgrade my Polycom to the newest firmware I could find (1.6.7). I am also trying to get provisioning working from a central server. I tired to reset with holding 468* down and it kept the settings the phone had on the phone. >From what I understand the settings on the phone override all. So I went into reset it from the phone and choose to format the firmware. Now when I try to boot it I am getting the following in the *-boot.log   0527180621|cfg  |4|00|Could not get all 512 bytes of the header. 0527181013|cfg  |4|00|Could not get all 512 bytes of the header. 0527181014|app1 |6|00|Error application is not present. 0527181014|app1 |6|00|Uploading boot log, time is SAT MAY 27 18:10:14 2006   I tried to put the old firmware and configs back in the directory but I get the same thing. Any help out there?   Thanks!   Curt 

 

 

Clint Neider

Email Administrator

[EMAIL PROTECTED]

Alta Resources | IT Application Services |
120 N Commercial St
| Neenah, WI
 54956 | Office (920)
751-5800 x 7472 |

 

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[asterisk-users] Re: Sip Trunks

[Martin Joseph]

On 2006-10-18 12:34:43 -0700, [EMAIL PROTECTED] said:


Hello, well, I need to configure two asterisk box like SIP trunks to send  sip
calls from one asterisk to the other and visceversa. So How I setup confi g
files to get this working?.Thanks.


You can do it via IAX2, there was a recipe posted here very recently 
that made this quite simple. Plus IAX2 saves bandwidth for trunked 
calls.


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


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[asterisk-users] Getting started with sample dial plans

[Mitch Miller]

Okay, I have Asterisk up and running on Fedora Core 5 with a TDM400 
board with one FXO and FXS module.  Zap is up and running and * is 
functioning with the modules.  Oh yeah, and I have some soft phones 
configured and have them working as well.


Now I'm ready to begin playing with dial plans and am having a difficult 
time getting started.


I'm looking for some simple samples that might demonstrate basic 
functionality such as running the "inside" phone extension when an 
incoming call is received.  Or, a simple "Dial 9 for an outside line" 
plan where whatever number is dialed (after the 9) is simply dialed via 
the Zap Line.


In other words, something that makes * nearly transparent to begin with.

Then, I'd like to slowly add to the dial plan as I learn more of the 
commands.


I have not found any good samples of dial plans (other than the defaults 
 built with *) that demonstrate basics like this.


Are there some online references I could work through?

I've been to Asterisk.com and their respective documentation, and 
Asteriskguru.com and not found what I'm looking for (maybe I overlooked 
it?).


Any guidance is sincerely appreciated.

-- Mitch

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Re: [asterisk-users] Embedded Asterisk

[Mike Diehl]

On Thursday 19 October 2006 14:10, Cory Andrews wrote:
> I caught a thread the other day concerning Astricon and users embedding
> Asterisk on a Linksys or Netgear broadband router.  I lost track of the
> email thread, if anyone is presently working with this scenario please
> shoot me an email.

I happen to know that November's Linux Journal will have an article about 
running Linux/Asterisk on a Linsys WRTGS54SL router.  Nothing too 
technical, but I hope you enjoy it.

Mike Diehl.
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[asterisk-users] /dev/zap/channel ownership

[Mitch Miller]

* is having permission problems accessing /dev/zap/channel.  When I 
look, these devices (everything in /dev/zap) shows root.root for uid and 
gid.  If I start Asterisk from the command line, it runs fine (running 
as Root).  When I start it as a service, I get


Oct 19 23:02:55 WARNING[10587] chan_iax2.c: Unable to open IAX timing 
interface: Permission denied
Oct 19 23:02:55 WARNING[10587] chan_zap.c: Unable to open 
'/dev/zap/channel': Permission denied
Oct 19 23:02:55 ERROR[10587] chan_zap.c: Unable to open channel 1: 
Permission denied


So ... I changed ownership on /dev/zap/* to asterisk.asterisk and now 
everything seems to be running fine.


My question is ... how would the ownership on these devices have 
changed?  (I've not yet rebooted, but I'm suspicious that they'll revert 
back to root.root).


-- Mitch
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Re: [asterisk-users] bristuff-0.3.0-PRE-1u for Asterisk 1.2.13 on junghanns downloads now

[Michiel van Baak]

On 02:39, Fri 20 Oct 06, Tzafrir Cohen wrote:
> On Thu, Oct 19, 2006 at 11:27:07PM +0200, Michiel van Baak wrote:
> > On 23:04, Thu 19 Oct 06, Vidar wrote:
> > > Bristuff has been updated;
> > > 
> > > http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1u.tar.gz
> > 
> > Thanks for the information.
> > 
> > It's a shame we need to read this here and not see it on
> > their website.
> 
> Also note that the changelog entry for 0.3.0-PRE-1u is missing from the
> CHANGES file. Nevertheless, that is a version for Asterisk 1.2.13 ,
> Zaptel 1.2.10 and libpri 1.2.4 .

Also note that the changelog mentioned -1t and that that
file is also available on FTP.
I think junghann.net needs a webmaster ;-)

I'm off to run diff to make the changelog myself...
-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer afficionados are both called users?"

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Re: [asterisk-users] question about CDR command

[William Piper]

I don't believe there is any quick & simple way of doing this. 
You would need to add the column in the DB and modify cdr.c.
 
I'm sure someone out there has a "step by step" doc on how to do this. You may try the #asterisk channel on irc.
 
bp 
On 10/19/06, unplug <[EMAIL PROTECTED]> wrote:
Thanks!!Just one more question.  Can I do the same "add fieldname=1" if I adda field "fieldname" in the cdr table to perform the same action?
On 10/19/06, William Piper <[EMAIL PROTECTED]> wrote:> In cdr_mysql.conf add "userfield=1" under the globals setting.>> bp
>>> On 10/18/06, unplug <[EMAIL PROTECTED]> wrote:> >> > I want to set some custom data in the field of userfield in table CDR> > as following.
> > exten => s,19,Set(CDR(userfield)=1234)> > exten => s,20,Dial(SIP/1234)> >> > However, the userfield doesn't get update after making the call.> > After that, I relocate the command as following.
> >> > exten => s,19,Dial(SIP/1234)> > exten => s,20,Set(CDR(userfield)=1234)> >> > The userfield doens't get update at all.  I don't know why the field> > can't update after issuing the command.  Anyone can help?
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Re: [asterisk-users] Embedded Asterisk

[Mike Diehl]

Doh!  Turns out it won't be November.  It will be a bit later.  Sorry.

On Thursday 19 October 2006 21:35, Mike Diehl wrote:
> On Thursday 19 October 2006 14:10, Cory Andrews wrote:
> > I caught a thread the other day concerning Astricon and users embedding
> > Asterisk on a Linksys or Netgear broadband router.  I lost track of the
> > email thread, if anyone is presently working with this scenario please
> > shoot me an email.
>
> I happen to know that November's Linux Journal will have an article about
> running Linux/Asterisk on a Linsys WRTGS54SL router.  Nothing too
> technical, but I hope you enjoy it.
>
> Mike Diehl.
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[asterisk-users] Re: Sip Trunks

[Martin Joseph]

On 2006-10-19 20:30:03 -0700, Martin Joseph <[EMAIL PROTECTED]> said:


On 2006-10-18 12:34:43 -0700, [EMAIL PROTECTED] said:


Hello, well, I need to configure two asterisk box like SIP trunks to se

nd  sip

calls from one asterisk to the other and visceversa. So How I setup con

fi g

files to get this working?.Thanks.


You can do it via IAX2, there was a recipe posted here very recently 
that made this quite simple. Plus IAX2 saves bandwidth for trunked 
calls.


Woops!  Wrong link, sorry...  Try 2.


http://astrecipes.net/index.php?n=204


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