[asterisk-users] Problems after upgrade asterisk

2006-07-19 Thread Iuri Gomes Diniz
Hi people,

  When a I upgrade my asterisk 1.2.4 to asterisk 1.2.9.1 or to asterisk
1.2.10, app_queue, after some time up, doesn't work (I think)

  When I call to the queue, the channels up:
Zap/1-1  [EMAIL PROTECTED]:4   Up  Queue(suporte3600)

  but nothing happens, the asterisk doesn't call any agent (agents are
dynamics and they are logged by agentcallbacklogin), i think so because
asterisk doesn't spawn a new channel of type LOCAL to call to the agent.

  My system is in production, so when this problem occurs, my E1 fill up
and I type 'show channels' on asterisk console, there are a lot of
channels executing app_queue like the line above, but after typed 'show
queues' nothing more happened, all commands don't do anything, asterisk
shows nothing.

All others functions works well, so when this problem occurs I can
call normally.

I have switched back to asterisk 1.2.4 because this version works well.

my system:
libpri 1.2.3
zaptel 1.2.7

pentium 4 3.0 GHZ with HT disabled on BIOS
TDM2400P with 24 FXS's 
TE205P with a ISDN E1.

all digium cards are in its own irq.

How I am doing the upgrade:

mv /usr/lib/asterisk /usr/lib/asterisk/old
cd asterisk-1.2.10
make install

I have tried 'make upgrade' too.
--
list of modules loaded is attached.

Thanks in advance, and sorry my bad english.
-- 
Iuri Gomes Diniz adm.iuri (at) digi.com.br
Network Admin and Programmer [http://clx.digi.com.br]
DIGINET [http://www.digi.com.br]
Natal - RN - Brazil.

Module Description  Use 
Count
res_musiconhold.so Music On Hold Resource   1
res_indications.so Indications Configuration0
res_monitor.so Call Monitoring Resource 1
res_adsi.soADSI Resource1
res_agi.so Asterisk Gateway Interface (AGI) 0
res_features.soCall Features Resource   1
res_config_odbc.so ODBC Configuration   1
res_odbc.soODBC Resource0
res_crypto.so  Cryptographic Digital Signatures 1
pbx_config.so  Text Extension Configuration 0
pbx_spool.so   Outgoing Spool Support   1
pbx_loopback.soLoopback Switch  1
pbx_realtime.soRealtime Switch  1
pbx_ael.so Asterisk Extension Language Compiler 0
pbx_functions.so   Builtin dialplan functions   0
chan_sip.soSession Initiation Protocol (SIP)1
chan_agent.so  Agent Proxy Channel  1
chan_mgcp.so   Media Gateway Control Protocol (MGCP)0
chan_iax2.so   Inter Asterisk eXchange (Ver 2)  0
chan_local.so  Local Proxy Channel  0
chan_features.so   Feature Proxy Channel0
chan_oss.soOSS Console Channel Driver   0
chan_phone.so  Linux Telephony API Support  0
chan_zap.soZapata Telephony w/PRI   9
app_dial.soDialing Application  1
app_playback.soSound File Playback Application  0
app_voicemail.so   Comedian Mail (Voicemail System) 0
app_directory.so   Extension Directory  0
app_mp3.so Silly MP3 Application0
app_system.so  Generic System() application 0
app_echo.soSimple Echo Application  0
app_record.so  Trivial Record Application   0
app_image.so   Image Transmission Application   0
app_url.so Send URL Applications0
app_disa.soDISA (Direct Inward System Access) Appli 0
app_adsiprog.soAsterisk ADSI Programming Application0
app_getcpeid.soGet ADSI CPE ID  0
app_milliwatt.so   Digital Milliwatt (mu-law) Test Applicat 0
app_zapateller.so  Block Telemarketers with Special Informa 0
app_setcallerid.so Set CallerID Application 0
app_festival.soSimple Festival Interface0
app_queue.so   True Call Queueing   2
app_senddtmf.soSend DTMF digits Application 0
app_parkandannounce.so Call Parking and Announce Application0
app_setcidname.so  Set CallerID Name0
app_lookupcidname.so   Look up CallerID Name from local databas 0

Re: [Asterisk-Users] CallerID popup

2006-02-03 Thread Iuri Gomes Diniz
On Fri, 3 Feb 2006 11:41:53 +0100
Giovanni Miano [EMAIL PROTECTED] wrote:

 Link event

For me, Link event only occurs when the called number pickup the call.

I prefer 'Newchannel' event when the 'State' are equal to 'Ringing'

-- 
Iuri Gomes Diniz adm.iuri (at) digi.com.br
Network Admin and Programmer [http://clx.digi.com.br]
DIGINET [http://www.digi.com.br]
Natal - RN - Brazil.



-- 
Iuri Gomes Diniz adm.iuri (at) digi.com.br
Network Admin and Programmer [http://clx.digi.com.br]
DIGINET [http://www.digi.com.br]
Natal - RN - Brazil.


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Re: [Asterisk-Users] How to have a phone ring another extension as soon as off-hook?

2006-01-24 Thread Iuri Gomes Diniz
From asterisk handbook-draft for zap:

;we can create a 'hotline' phone by placing a
;phone in a special context
;and setting it to answer immediately. In
;extensions.conf we can route
;the phone to an IVR, direct to security, or
;make it call Steak-Out

context = hotline
immediate = yes
channel = 24
Page 55


for sip phones or ATA's, The Grandstream phones have an option called off-hook 
autodial

On Fri, 20 Jan 2006 12:32:32 -0500
Script Head [EMAIL PROTECTED] wrote:

 I am seeking to implement the following behavor:
 
 When a headset on phone1 is picked up, phone2 rings right away, without any
 need to dial numbers on phone1. Is this possible to implement?
 
 ScriptHead


-- 
Iuri Gomes Diniz adm.iuri (at) digi.com.br
Network Admin and Programmer [http://clx.digi.com.br]
DIGINET [http://www.digi.com.br]
Natal - RN - Brazil.


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Re: [Asterisk-Users] TE110P + PRI incoming + outgoing extensions question

2006-01-21 Thread Iuri Gomes Diniz
On Sat, 21 Jan 2006 08:32:26 -0500
Doug Lytle [EMAIL PROTECTED] wrote:

 Why would an incoming call have a destination of 1153?

On my asterisk, when a call comes from E1 the default destination is the last 4 
digits.


-- 
Iuri Gomes Diniz adm.iuri (at) digi.com.br
Network Admin and Programmer [http://clx.digi.com.br]
DIGINET [http://www.digi.com.br]
Natal - RN - Brazil.


-- 
Esta mensagem foi verificada pelo sistema de anti-virus e
 acredita-se estar livre de perigo.

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