Joao Pereira
--
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt
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Hello
I have a 4 span PRI board with Zaptel, and Im using it for a long time.
In the last days I noticed that the result of "zap show status" show a
ZTDUMMY but I never installed it:
o*CLI> zap show status
Description Alarms IRQ
bpviol CRC4
T4XXP
Em 17-03-2010 20:28, Doug Lytle escreveu:
> Joao Gomes Pereira wrote:
>
>> What could be missing?
>>
>>
>>
> Running ldconfig as root
>
>
>
>
Thanks, thats it!!!
Now the module is loaded.
I just hope the FAX code works:
[macro-faxr
Em 17-03-2010 20:51, VinÃcius Fontes escreveu:
> - "Joao Gomes Pereira" escreveu:
>
>
>> Hello
>> Im trying to receive FAXes with my Asterisk with "rxfax" command.
>>
>> To do that, Im trying to load the "app_fax.so" module.
"/usr/local/lib" is in my "ld.so.conf":
cat /etc/ld.so.conf
include ld.so.conf.d/*.conf
/etc/ld.so.conf.d/*.conf
/usr/local/lib
/usr/include
/usr/local/include
What could be missing?
Thanks
Regards
Joao Pereira
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Joao Go
Hello
I'm trying to install Asterisk in a Linux server without compiler, yum
or apt-get.
Is this possible? Where can I find the pre compiled binaries?
Thanks
Regards
Joao Pereira
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Hello
Im configuring an asterisk peer, wich uses port 5060 to send and port
5061 to receive signaling.
So, wich port should I put in my asterisk SIP trunk configuration?
port = 5060
or
port = 5061
?
Thanks
Regards
Joao Pereira
--
_
Hello
Im configuring an asterisk peer, wich uses port 5060 to send and port
5061 to receive signaling.
So, wich port should I put in my asterisk SIP trunk configuration?
port = 5060
or
port = 5061
?
Thanks
Regards
Joao Pereira
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.@default:1Up
ChanSpy(Agent)
(the agent is using extension 211)
I have more then 10 lines like these. Why do the ChanSpy calls dont hang up?
Thanks
Regards
Joao Pereira
--
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@s
o Pereira
--
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt
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Nice!!!
Thanks a lot.
Its not the case (because Im using a fixed IP), but if the IP where dynamic?
Thanks
Regards
Joao Pereira
Randy R wrote:
> externip=123.123.123.123
>
> On Tue, Dec 1, 2009 at 4:32 PM, Joao Gomes Pereira
> wrote:
>
>> Hello
>> I'm trying
can I force Asterisk to register with its public IP?
Is it possible to configure STUN in an Asterisk trunk?
Thanks
Regards
Joao Pereira
--
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt
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I had to restart Asterisk, and now the module is loaded.
Thanks a lot for the help
Joao Pereira
--
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt
Danny Nicholas wrote:
> Try "module load app_chanspy.so" from CLI. If that
rds
Joao Pereira
Danny Nicholas wrote:
> App_chanspy
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joao Gomes
> Pereira
> Sent: Thursday, October 22, 2009 10:08 AM
> To: asterisk-
(default, 555, 1)
It seems that Asterisk doesn't have ChanSpy enabled... is this possible?
Which Asterisk module do I have to enable?
Thanks
Regards
Joao Pereira
--
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: go
-
From: Joao Gomes Pereira [mailto:gomespere...@startel.pt]
Sent: Thursday, September 03, 2009 11:56 AM
To: Danny Nicholas; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dahdi configuraion / error
it looks like this:
tail /etc/dahdi/modules
# Autogenerated
catumbela kernel: Completed startup!
Sep 3 18:02:39 catumbela dahdi: Running dahdi_cfg: succeeded
Joao Gomes Pereira escreveu:
> it looks like this:
>
> tail /etc/dahdi/modules
> # Autogenerated by /usr/sbin/dahdi_genconf (Dahdi::Config::Gen::Modules)
> on Wed Jun 24 12:41:26
;No such device" is sometimes an indication that /etc/init.d/dahdi start did
> not load the driver.
> What does /etc/dahdi/modules look like?
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On
FIG failed on span 1: No such device or address (6)
[r...@catumbela modules]#
Any idea of what could be nmissing?
Thanks a lot
Joao Pereira
--
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt
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SPAN 1: Primary Sync Source
VPM400: Not Present
VPM450: Not Present
Completed startup!
DMESG
Could this be a configuration issue or a hardware problem?
Thanks
regards
Joao Pereira
--
StarTel - A Rede Livre
catumbela kernel: TE4XXP: Version Synchronization Error!
What could be worng with my dahdi configuration?
Thanks
regards
Joao Pereira
Joao Gomes Pereira escreveu:
> Hello to all
> I'm using asterisk 1.4 and dahdi.
> I had everything working fine, and I could place calls through my
cas=1-15:1101
cas=17-30:1101
dchan=16
---
What could be the problem?
Why was this working fine and now the channel is RED?
Thanks
Regards
Joao Pereira
--
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomes
--
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
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ks
regards
Joao Pereira
--
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Reg
Status
kamailio/my_username xxx.xxx.xxx.xxx 5060 OK
(890 ms)
Is there something missing in my SIP.CONF to improve the compatibility
with Kamailio?
How can I debug the RTP stream in Asterisk?
Thanks
Regards
Joao Pereira
--
StarTel - A Rede Livre
Joao Gomes Pereira
www.st
rt Status
kamailio/my_username xxx.xxx.xxx.xxx 5060 OK
(890 ms)
Is there something missing in my SIP.CONF to improve the compatibility
with Kamailio?
How can I debug the RTP stream in Asterisk?
Thanks
Regards
Joao Pereira
--
StarTel - A Rede Livre
Joao Gomes Pereira
w
Pereira
--
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt
Tom Poe wrote:
> Hello: I have the linux version 2.0 of x-lite downloaded. Does anyone
> know exactly what settings needed to reach the asterisk server on my
> hom
available) agents I have in
a queue before sending a call?
Thanks
Regards
Joao Pereira
--
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt
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StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt
Geraint Lee wrote:
> Take a look at:
> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Random
>
> You should be able to do what you want with this, it obviously won'
wer
; 2 in 3 calls go to queue_1
exten => 123,x,Queue(queue_1)
; 1 in 3 calls go to queue_2
exten => 123,x,Queue(queue_2)
But how can I configure this call distribution?
Thanks
Regards
Joao Pereira
--
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomesper
the queues.conf file. Do try.
>
> Regards,
>
> Kurian Thayil.
>
> On Sun, 2009-06-07 at 17:51 +0100, Joao Gomes Pereira wrote:
>
>> Hello to all
>> I'm trying to record the calls going to my queues, but asterisk creates
>> 2 files, one with the inbou
=1
wrapuptime=19
ackcall=no
group=1
agent => 600,1234,Jose
agent => 601,1234,Maria
Thanks
Regards
Joao Pereira
--
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt
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