Hello,
I have a 3 server asterisk configuration where one asterisk (say A) (v
1.4.25) has a digiuim card connected to E1 from which calls are routed
to another asterisk server (B) (1.6.0.9) over IAX trunk from which
calls get routed to third server (C) (1.6.0.9) again via IAX trunk.
SIP clients
On Aug 13, 2009, at 6:24 PM, Usman Tahir wrote:
Hi Raimund,
Hello,
snom uses basically the same concept. As explained under:
http://wiki.snom.com/Settings/user_failover_identity.
You select the line id that should be used when a registration fails.
thank you for the link, as a matter of
Asterisk version 1.4
From: Neeraj Chand
Sent: Friday, 14 August 2009 8:17 PM
To: 'asterisk-users@lists.digium.com'
Subject: [asterisk-users] Time of Day Routing
Hi David,
With this:
ifTime(00:00-12:00|*|*|*)
Whatever time you specify at the end,
Hi David,
With this:
ifTime(00:00-12:00|*|*|*)
Whatever time you specify at the end, I believe asterisk continues to
evaluate this condition as true for 2 more minutes.
So in this case, it will be valid for 00:00-12:02, even though you've
specified 12:00
Cheers!
Neeraj
On 14 Aug 2009, at 09:17, Neeraj Chand wrote:
Asterisk version 1.4
From: Neeraj Chand
Sent: Friday, 14 August 2009 8:17 PM
To: 'asterisk-users@lists.digium.com'
Subject: [asterisk-users] Time of Day Routing
Hi David,
With this:
ifTime(00:00-12:00|*|*|*)
Whatever time you specify
Hi folks,
Going to astricon this year? Feeling a bit nervous as planning to take
the exam this time. Any one else doing the same?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15
Hi,
Can I play a prompt after hanging up a call? I have tried below but failed.
...
exten = s,n,Dial(SIP/1234)
...
exten = h,1,Playback(demo-instruct)
-- Executing [...@macro-safedial:2] Playback(SIP/3601-09856bf0,
demo-instruct) in new stack
[Aug 14 17:24:03] WARNING[2496]:
Hello all,
I'm trying to conect two asterisk servers using two B410p Digium
cards. One card on each server. I just setting up the first BRI port
on server A as nt_ptp and the first BRI port on server B as te_ptp.
I use an ethernet wire to connect the first port of server A (nt_ptp)
with the first
On Friday, August 14, 2009, Rilawich Ango wrote:
Hi,
Can I play a prompt after hanging up a call? I have tried below but
failed.
...
exten = s,n,Dial(SIP/1234)
...
exten = h,1,Playback(demo-instruct)
-- Executing [...@macro-safedial:2] Playback(SIP/3601-09856bf0,
demo-instruct) in
In article 05d03313-994b-4892-b045-f61332ddb...@geekinter.net,
Steve Howes st...@geekinter.net wrote:
On 14 Aug 2009, at 09:17, Neeraj Chand wrote:
Asterisk version 1.4
From: Neeraj Chand
Sent: Friday, 14 August 2009 8:17 PM
To: 'asterisk-users@lists.digium.com'
Subject:
Tony Mountifield wrote:
Hmm, I would still consider it a bug, whether on 1 or 2 minute resolution.
I haven't seen the 2 minutes issue with the below:
GotoIfTime(07:59-16:59|mon-fri|*|*?office-hours,s,1)
Our plant closes at 5pm. And, at exactly 5pm, the afterhours context
takes over.
Tony Mountifield wrote:
In article 05d03313-994b-4892-b045-f61332ddb...@geekinter.net,
Steve Howes st...@geekinter.net wrote:
On 14 Aug 2009, at 09:17, Neeraj Chand wrote:
Asterisk version 1.4
From: Neeraj Chand
Sent: Friday, 14 August 2009 8:17 PM
To:
Tony Mountifield wrote:
In article 05d03313-994b-4892-b045-f61332ddb...@geekinter.net,
Steve Howes st...@geekinter.net wrote:
On 14 Aug 2009, at 09:17, Neeraj Chand wrote:
Asterisk version 1.4
From: Neeraj Chand
Sent: Friday, 14 August 2009 8:17 PM
To:
In article 4a855630.5080...@arcdiv.com, SIP s...@arcdiv.com wrote:
Tony Mountifield wrote:
In article 05d03313-994b-4892-b045-f61332ddb...@geekinter.net,
Steve Howes st...@geekinter.net wrote:
On 14 Aug 2009, at 09:17, Neeraj Chand wrote:
Asterisk version 1.4
From: Neeraj
On Friday 14 August 2009 08:15:55 Tony Mountifield wrote:
In article 4a855630.5080...@arcdiv.com, SIP s...@arcdiv.com wrote:
Tony Mountifield wrote:
In article 05d03313-994b-4892-b045-f61332ddb...@geekinter.net,
Steve Howes st...@geekinter.net wrote:
On 14 Aug 2009, at 09:17, Neeraj
hi all
do you guys know why asterisk sometimes, in the cdr put the dst (the
extension) number in the src ??
I have 4 digit extensions (DID) (95XX) and sometimes, the same values if found
in the src that usually have the calling user caller id.
Example Q-aereos.csv
Hi all,
I hope you guys can help me out. I got a problem with using function
CURL. I
did Set(CURL=${CURL(URL)}); but the URL I was using is https, so when I
generated the call, the CURL function could not get access to that
https://URL server. What should I do with it? Thank you very much
Hi, I have an asterisk connected with PRI (Zap channels).
If I try to call a number, and recieve cause code 27 because the line 553192
is out of service, but the call continue...is it ok?
Here the console messages
-- Executing [...@troncal-pri-76:5] Dial(Zap/1-1, Zap/g1/553192) in
new stack
how do i troubleshoot no ring tone. It was working and all i added was the
lines below now it doesn't ring.
Edit sip_nat.conf for proper NAT:
localnet=192.168.1.0/255.255.255.0 externhost=pbx.DOMAIN.com (Set your external
hostname name here)
externrefresh=10
fromdomain=DOMAIN.com (Set your
On 14 Aug 2009, at 15:01, Dpto. de Sistemas wrote:
hi all
do you guys know why asterisk sometimes, in the cdr put the dst (the
extension) number in the src ??
I have 4 digit extensions (DID) (95XX) and sometimes, the same
values if found
in the src that usually have the calling user caller
Hi
I just solve my problem today. Just a package on redhat that I need install.
H.
___
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
Hello,
I never use externhost
y use \
externip=public ip
And work fine
Regards
On Fri, Aug 14, 2009 at 9:18 AM, Ott Rose sixfourimp...@hotmail.com wrote:
how do i troubleshoot no ring tone. It was working and all i added was the
lines below now it doesn't ring.
Edit sip_nat.conf for
On 14 Aug 2009, at 15:18, Ott Rose wrote:
how do i troubleshoot no ring tone. It was working and all i added
was the lines below now it doesn't ring.
Edit sip_nat.conf for proper NAT:
localnet=192.168.1.0/255.255.255.0 externhost=pbx.DOMAIN.com (Set
your external hostname name here)
On Friday 14 August 2009 09:04:12 Wenbin Zhang wrote:
Hi all,
I hope you guys can help me out. I got a problem with using function
CURL. I
did Set(CURL=${CURL(URL)}); but the URL I was using is https, so when I
generated the call, the CURL function could not get access to that
i call to my tollfree number buy my CLI send the next error:
Aug 14 08:15:22 WARNING[25931]: format_wav.c:169 check_header: Unexpected
freqency 22050
Aug 14 08:15:22 WARNING[25931]: file.c:441 ast_filehelper: Unable to open
file on /var/lib/asterisk/sounds/procall3.wav
Aug 14 08:15:22
Playback is expecting a frequency of 8000. use sox to correct. As for
101/103, that is how the dialplan is written, not an error per se.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bayardo
Sanchez
Sent: Friday, August
the audio is in format wav i save in Format PCM attributte 8,000 KHz; 8bit;
Mono 7kb/s
my extension.conf is the next:
exten = 651085,1,Playback(procall3)
exten = 651085,n,Playback(procall3)
exten = 651085,n,Queue(procall|n|||)
exten = 651085,n,Playback(voicemail-invitation)
exten
Did you get CDRTool to work with Asterisk or Areski's CDR Stats?
On Fri, Aug 14, 2009 at 10:20 AM, harry R rhm.noa...@gmail.com wrote:
Hi
I just solve my problem today. Just a package on redhat that I need
install.
H.
___
-- Bandwidth and
Tilghman Lesher wrote:
On Friday 14 August 2009 09:04:12 Wenbin Zhang wrote:
Hi all,
I hope you guys can help me out. I got a problem with using function
CURL. I
did Set(CURL=${CURL(URL)}); but the URL I was using is https, so when I
generated the call, the CURL function could not
You probably want to set the option
CURLOPT_SSL_VERIFYPEER to FALSE.
Especially with chained certificates (cheapos from godaddy, etc), I have had
lots of trouble with CURL being able to validate a cert. That's probably
because I didn't tell it where the root certs were... but either way.
yes i just copied that form the freepbx site. sorry about that
From: st...@geekinter.net
To: asterisk-users@lists.digium.com
Date: Fri, 14 Aug 2009 15:43:00 +0100
Subject: Re: [asterisk-users] no ring tone
On 14 Aug 2009, at 15:18, Ott Rose wrote:
how do i troubleshoot no ring tone.
i changed it and still didn't ring. however it did ring on one call to a cell
phone but it hasn't done it again.
Date: Fri, 14 Aug 2009 09:39:33 -0500
From: crt.ro...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] no ring tone
Hello,
I never use externhost
y use
David Gibbons wrote:
You probably want to set the option
CURLOPT_SSL_VERIFYPEER to FALSE.
Especially with chained certificates (cheapos from godaddy, etc), I have had
lots of trouble with CURL being able to validate a cert. That's probably
because I didn't tell it where the root certs
The error is here,
Q-aereos,1147739512,9536,outbound,1147739512,DAHDI/
30-1,SIP/9536-137a4cc0,Dial,SIP/9536|60|t,2009-08-13
13:41:03,2009-08-13 13:41:06,2009-08-13 13:44:01,
178,175,ANSWERED,DOCUMENTATION,1250170815.1512,
9536 is extension and 1147739512 is dst,
Why SOMETIMES cdr put the dst
Using asterisk 1.6.2.1 and dahdi 2.2.0.2. dadhi-linux installed just
fine. Using dahdi_dummy as there is no card in system. Did not install
libpri, again, no card.
When compiling asterisk, I include -with-dahdi and everything
./configure's fine but when I do 'make', everything goes fine but
Procall3 is /var/lib/asterisk/sounds/procall3.wav?
IMO, procall should look like this:
[procall]
exten = s,1,Set(TIMEOUT(digit)=7) ;
exten = s,2,Set(TIMEOUT(response)=10)
exten = s,3,Set(CHANNEL(language)=en) ; define como idioma predefinido el
ingles y usas las voces en este idioma
exten =
Boehm, Matthew wrote:
Using asterisk 1.6.2.1 and dahdi 2.2.0.2. dadhi-linux installed just
fine. Using dahdi_dummy as there is no card in system. Did not install
libpri, again, no card.
There isn't any 1.6.2.1 release of Asterisk; which version did you try
to build?
--
Kevin P. Fleming
Sorry. Fat-finger. 1.6.1.2. Found out that --with-tonezone is required
when ./configure'ing asterisk in order to get chan_dahdi to compile.
Could that be documented somewhere? Or auto checked in ./configure if
--with-dahdi is specified? Wasted over an hour trying to figure this
out.
Thanks,
yes procall3 is in /var/lib/asterisk/sounds/procall3/wav
erase these:
exten = i,1,Playback(invalid)
exten = i,2,Playback(goodbye)
exten = i,3,hangup
exten = t,1,goto(procall,s,1)
exten = h,1,Hangup
?
On Fri, Aug 14, 2009 at 9:56 AM, Danny Nicholas da...@debsinc.com wrote:
Procall3 is
Dpto. de Sistemas escribió:
The error is here,
Q-aereos,1147739512,9536,outbound,1147739512,DAHDI/
30-1,SIP/9536-137a4cc0,Dial,SIP/9536|60|t,2009-08-13
13:41:03,2009-08-13 13:41:06,2009-08-13 13:44:01,
178,175,ANSWERED,DOCUMENTATION,1250170815.1512,
9536 is extension and 1147739512 is
Dear All,
I am trying to using E1 PRI Connection with vicidialnow setup..Calls
are landed in asterisk .From Asterisk CLI i can see the caller id from where
the call came ..but the calls are rejects.
So what should i do now for forwarding this call to a agent who is using
Hello, i got the same problem with a Digium Card an AEX808E, with dahdi
linux 2.2.0 dahdi tools 2.2.0 and asterisk 1.6.1.4
i did update to the latest of everything hoping it will fix the problem,
but it still remains.
i got:
Aug 14 02:29:16 ctg01 kernel: [ 9257.702038] wctdm24xxp0: Missed
This would have to do with the reason they're being rejected, wouldn't
it? What is the reason?
Tareq Kibria wrote:
Dear All,
I am trying to using E1 PRI Connection with vicidialnow
setup..Calls are landed in asterisk .From Asterisk CLI i can see the
caller id from where the
Boehm, Matthew wrote:
Sorry. Fat-finger. 1.6.1.2. Found out that --with-tonezone is required
when ./configure'ing asterisk in order to get chan_dahdi to compile.
Could that be documented somewhere? Or auto checked in ./configure if
--with-dahdi is specified? Wasted over an hour trying to
Hello
One question
In sip.con or sip_additionals.conf, in freepbx, the context of your client
do you put
nat = yes
externip =
You put your public ip.
Are you sure that?
Regards
On Fri, Aug 14, 2009 at 10:43 AM, Ott Rose sixfourimp...@hotmail.comwrote:
i changed it and still didn't
No. dont erase them. You need to move procall3.wav to
/var/lib/asterisk/sounds/custom/procall3.wav. The nickel tour of procall
[procall]
exten = s,1,Set(TIMEOUT(digit)=7) ;
Set timeout for digits in background (4)
exten = s,2,Set(TIMEOUT(response)=10)
Set timeout for response in bg (4)
Dear all, does anybody know about a complete set of neutral Spanish sounds
to use in my Asterisk voicemail ???
Because when I get a Spanish sounds package, it always is incomplete.
I live in Argentina, so I prefer neutral voices.
Special thanks
Alejandro
i am not sure what you are talking about. i have extensions and my sip trunk
config in that file. see below
[200]
deny=0.0.0.0/0.0.0.0
type=friend
secret=200
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
mailbox=...@device
host=dynamic
dtmfmode=rfc2833
dial=SIP/200
Received this on the console
-- IAX2/76.21.238.129:4569-4986 requested special control 20, passing it
to SIP/magicjack-08225a58
Did a Google search, but reached a dead end
Can anyone explain?
Something need to be changed in my configuration?
The call completed satisfactorily.
Inbound IAX trunk
I assume that the supplied voice for Spanish is cepstal - marta
(www.cepstral.com http://www.cepstral.com/ ). In my (English)
installation, there are about 1800 files, so I can see how/why the set might
be incomplete.
_
From: asterisk-users-boun...@lists.digium.com
Try http://www.voipnovatos.es/voces http://www.voipnovatos.es/voces
Un saludo
Enrique Mora
Context M.I.S. SL
em...@context.es
logo-email
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] En nombre de Alejandro
Cabrera Obed
Enviado
On Friday 14 August 2009 10:48:08 Wenbin Zhang wrote:
David Gibbons wrote:
You probably want to set the option
CURLOPT_SSL_VERIFYPEER to FALSE.
Especially with chained certificates (cheapos from godaddy, etc), I have
had lots of trouble with CURL being able to validate a cert. That's
Tilghman Lesher wrote:
On Friday 14 August 2009 10:48:08 Wenbin Zhang wrote:
David Gibbons wrote:
You probably want to set the option
CURLOPT_SSL_VERIFYPEER to FALSE.
Especially with chained certificates (cheapos from godaddy, etc), I have
had lots of trouble with CURL being able
You could do a System(wget xx)...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Wenbin Zhang
Sent: Friday, August 14, 2009 3:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Is it possible to place multiple phone numbers in a single outbound
.call file? If I try doing this, only the last phone number in the file
is called.
However, if I use 1 file per phone number, then Asterisk attempts to
process all generated CALL files at once, incrementing the retry count
for
Danny Nicholas wrote:
You could do a System(wget xx)...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Wenbin Zhang
Sent: Friday, August 14, 2009 3:09 PM
To: Asterisk Users Mailing List - Non-Commercial
There are at least two solutions to this dilemma:
#1. dial using a context instead of a number
Or
#2 when creating the call files, use a future time (I use 60 seconds plus
the expected length of the call). The PERL module Asterisk::Outgoing is
supposed to handle it, but I had to modify mine
This gets into a bit of a hack, but you could do a quick-and-dirty AGI to
do the WGET then set the variable based on what you got back. If you did it
in perl, you could actually use LWP instead of WGET.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
On Friday 14 August 2009 15:09:27 Wenbin Zhang wrote:
Tilghman Lesher wrote:
On Friday 14 August 2009 10:48:08 Wenbin Zhang wrote:
David Gibbons wrote:
You probably want to set the option
CURLOPT_SSL_VERIFYPEER to FALSE.
Especially with chained certificates (cheapos from godaddy,
On Friday 14 August 2009 15:48:08 Tilghman Lesher wrote:
On Friday 14 August 2009 15:09:27 Wenbin Zhang wrote:
Tilghman Lesher wrote:
On Friday 14 August 2009 10:48:08 Wenbin Zhang wrote:
David Gibbons wrote:
You probably want to set the option
CURLOPT_SSL_VERIFYPEER to FALSE.
Tilghman Lesher wrote:
On Friday 14 August 2009 15:48:08 Tilghman Lesher wrote:
On Friday 14 August 2009 15:09:27 Wenbin Zhang wrote:
Tilghman Lesher wrote:
On Friday 14 August 2009 10:48:08 Wenbin Zhang wrote:
David Gibbons wrote:
You probably want to
Tilghman Lesher wrote:
On Friday 14 August 2009 15:48:08 Tilghman Lesher wrote:
On Friday 14 August 2009 15:09:27 Wenbin Zhang wrote:
Tilghman Lesher wrote:
On Friday 14 August 2009 10:48:08 Wenbin Zhang wrote:
David Gibbons wrote:
You probably want to
On Friday 14 August 2009 16:14:32 Wenbin Zhang wrote:
Tilghman Lesher wrote:
On Friday 14 August 2009 15:48:08 Tilghman Lesher wrote:
On Friday 14 August 2009 15:09:27 Wenbin Zhang wrote:
Tilghman Lesher wrote:
On Friday 14 August 2009 10:48:08 Wenbin Zhang wrote:
David Gibbons wrote:
Use a standard network cable - but you have to activate the 'terminate'
jumper on the NT end.
- Also, the new BRI stuff in dahdi is much easier to work with than misdn.
PaulH
voip crazy wrote:
Hello all,
I'm trying to conect two asterisk servers using two B410p Digium
cards. One card on
Tilghman Lesher wrote:
Regardless of how you think it should work, the poster above described
precisely the way it works. If your end boundary is 12:00, it will evaluate
as true all the way up until 12:01:59. If you don't want that, another poster
has suggested using 11:59, which will work
What files at a bare minimum need to be in /etc/asterisk for an
asterisk server that does sip only and voicemail. I'm setting up an
asterisk server to provide service for a single SIP softphone
extension with SIP origination and termination. The main purpose of
using * is for voicemail and
On Fri, 14 Aug 2009, Eric Fort wrote:
What files at a bare minimum need to be in /etc/asterisk for an asterisk
server that does sip only and voicemail. I'm setting up an asterisk
server to provide service for a single SIP softphone extension with SIP
origination and termination. The main
On 15/08/09 2:20 PM, Eric Fort wrote:
What files at a bare minimum need to be in /etc/asterisk for an
asterisk server that does sip only and voicemail. I'm setting up an
asterisk server to provide service for a single SIP softphone
extension with SIP origination and termination. The main
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