[Asterisk-Users] ISDN cards using CAPI interface

[Jesus Bermudez Riquelme - Pcmur Soluciones Informaticas]

Hi all,
 
if you configure a passive ISDN card with CAPI 
support, does it mean that can be used with asterisk (using chan-capi, of 
course) ??
 
Thanks
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Re: [Asterisk-Users] Master Telephone

[Jesus Bermudez Riquelme - Pcmur Soluciones Informaticas]

First, you have to configure your zapata.conf & 
sip.conf to support your hardware (see http://www.voip-info.org/wiki/index.php?page=Asterisk and 
read 
http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip ,this 
last is a must-read one)
 
After that, you have to see if all incoming calls 
are done within work hours:
 
[incoming]
 
exten => 
s,1,GotoIfTime(9:00-15:00,mon-fri,*,*?accepted,1,10)
exten => 
s,2,thefunctionyouwantdoneifisnthourtime
 
[accepted]
 
exten => 
1,1,Dial(thephoneyouwantasmaster)
; add here what you want if that phone is in 
use
 
as you see, you have no need for new hardware and 
programming is really simple ;)
 
 

  - Original Message - 
  From: 
  Johnathan 
  Falk 
  To: Asterisk-Users@lists.digium.com 
  
  Sent: Tuesday, November 22, 2005 4:22 
  PM
  Subject: [Asterisk-Users] Master 
  Telephone
  
  
  I am the network administrator for 
  a small school in Michigan.  We are currently using an 
  older proprietary pbx system and are trying very hard to get away from this 
  one vender lock in. I have set up an asterisk server using the version 1.2 of 
  asterisk.  Our current system has a master telephone used by the head 
  secretary that can transfer anyone’s calls and just generally handle all phone 
  redirection.  Kind of like a head receptionist.  The one rule we 
  have about our telephone system is that during school hours a person must 
  answer.  Our superintendent refuses to have a machine answer if there are 
  people working.  So the head secretary must either redirect the calls to 
  someone who is there or take the call herself.  How can we accomplish 
  this with asterisk? Also if it can’t be accomplished through programming what 
  special hardware would we be required to 
purchase.
   
  Johnathan FalkNetwork 
  Administrator
  Clinton Community 
  Schools
  1-517-442-9622
  
  

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[Asterisk-Users] Problem with SIP channels

[Jesus Bermudez Riquelme - Pcmur Soluciones Informaticas]

Hi all,
i've a problem in my Asterisk system. We have 
around 30 SIP phones connected to an asterisk system, and sometimes some SIP 
channel (associated to an extension) gets busy all the time, even when that 
extension isn't in use.
 
We have a workaround for this, as we can't restart 
asterisk in work hours, we assign other extension to that phone and create an 
alias for calling there.
 
When asterisk is restarted, all extensions aswer 
the way it's supposed to be.
 
Any clues?
 
Thank you :)
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Re: [Asterisk-Users] Hung Zap channels

[Jesus Bermudez Riquelme - Pcmur Soluciones Informaticas]

Hmm, it sounds to me the hangup detection problem. 
Here's a way to solve that problem:
Apply this patch:http://www.maxosystem.net/asterisk/asterisk-stable-polarity-v5.diff
$ cd /usr/src/asterisk/channels$ patch 
chan_zap.c < /your/route/here/asterisk-stable-polarity-v5.diff
 
and in zapata.conf :
 
answeronpolarityswitch=yeshanguponpolarityswitch=yes
 

Sure, it will only help if your PSTN company does hangup detection by 
changing polarity of the line, refer to their line specificacions to more 
info.
 
Hope it helps ;)
 
 
- Original Message - 
From: "John Heng" <[EMAIL PROTECTED]>
To: 
Sent: Friday, November 18, 2005 1:56 
AM
Subject: [Asterisk-Users] Hung Zap 
channels
Hi all,I'm running asterisk 1.0.9 (yes I know - 1.2 has just been 
released) with a TDM400P board that has 4 FXO port. Once in a while, I've found 
that the zap channel will get stuck (or blocked) even after the call has ended. 
Sometimes this is when someone has left a voice msg, but not always. The 
way I've fix this is to issue a "soft hangup" command for that zap channel. 
However, I'm not always aware of this until a user tells (or complains to) me. 
What I would like to know is if there is a way to reset all the zap 
channels or re-initialize the drivers without restarting Asterisk. If so, I 
could set up a cron job to do it once or twice a week, in the middle of the 
night. Any suggestion guys??CheersJ 
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Re: [Asterisk-Users] Hangup detection - TDM400P

[Jesus Bermudez Riquelme - Pcmur Soluciones Informaticas]

Here in Spain we had that problem since the hangup here is done by changing 
line polarity.

It is solved by aplying this patch:

http://www.maxosystem.net/asterisk/asterisk-stable-polarity-v5.diff
$ cd /usr/src/asterisk/channels
$ patch chan_zap.c < /your/route/here/asterisk-stable-polarity-v5.diffand in 
zapata.conf :answeronpolarityswitch=yes
hanguponpolarityswitch=yesHope it helps ;)- Original Message - 
From: "Marco Supino" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Thursday, November 17, 2005 5:20 PM
Subject: Re: [Asterisk-Users] Hangup detection - TDM400P



Yes, didnt change anything

Marco.


Angelito Manansala wrote:

hmmm
di you try this ;hanguponpolarityswitch=yes

Cheerz!

On 11/17/05, Marco Supino <[EMAIL PROTECTED]> wrote:


Hi,

I have a long delay when detecting hangups on the TDM400P card, with 4
FXO ports,

When an incoming call dial's in, when hanging up, the asterisk will
detect the hangup only after 10 seconds, i searched around, and found
many similar problems, but no solution, i tried some options in
zapate.conf , but nothing helped, any solution ?

the lines are coming from SBC in San Fransisco, i asked them if i have
"disconnect supervision", and they said i do have it.

Marco.

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--
Best Regards,
Angelito Manansala
www.voicefidelity.net
Mobile: +639175425807
DID: (+63) 44 7906770
msn: [EMAIL PROTECTED]
skype: bulcrack
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