On Thursday 24 March 2011 04:50:48 Jonas Kellens wrote:
> On 03/24/2011 10:45 AM, Rizwan Hisham wrote:
> > You have to use adaptive cdr for this functionality. In 1.8 the conf
> > file for adaptive cdr is cdr_adaptive_odbc.conf. The sample conf file
> > should tell you everything.
> >
> > If you a
On Thursday 24 March 2011 11:38:54 Gordon Henderson wrote:
> On Wed, 23 Mar 2011, Douglas Mortensen wrote:
> > 1.2? 1.4? 1.6? 1.8?
>
> 1.2 has been the most stable version for me.
>
> Same setups with 1.4 +DAHDI has never been as stable with random crashes
> and re-starts - however they're not pr
That sounds good but, i would like it the other way arround. I have over 90
extensions that are NOT allowed to use the trunk, and 2 that are.. So
blacklisting everything will take for ever ;D.
On Thu, Mar 24, 2011 at 10:01 PM, Danny Nicholas wrote:
> Just use “Ex-girlfriend” logic on your dial
So make a whitelist
What I do is create a outbound route with the allowed cid and then have another
route which goes to a not allowed recording which catches all other caller Id's
-Original Message-
From: Peter den Hartog
Sender: asterisk-users-boun...@lists.digium.com
Date: Fri, 25 Mar
On Thursday 24 March 2011 12:02:38 vip killa wrote:
> If you are new to VoIP, you are better off learning FreeSWITCH
And if you're new to analog recordings, you're better off purchasing
Sony BetaMax. How is your BetaMax deck, btw?
--
Tilghman
--
Hi
Using ${EXTEN:0:3}
will only return the first 3 digits entered
Ish
On Wed, 2011-03-23 at 16:27 -0400, Eddie Mikell wrote:
> All:
>
> Some of the people who dial into to our system will press the pound key
> when entering an extension for the directory key. When waitexten gets
> that, I g
On Thu, 2011-03-24 at 21:58 +0100, Thomas Winter wrote:
> Hi list,
> I have an 44100 Hz file with human voice, stereo with 16Bit.
> When convertig this to 8 kHz, mono I loose a lot of quality and have
> some ground noise. I tried several sox options but without success.
> Can somebody help
>
Hi Olivier,
here is solutions for your situation , ideally you need to talk with
Provider and they can set SIP URI
for given DID numbre , but that can be solved by dial-plan like this.
exten => _003318364,1,Set(foo=${SIP_HEADER(To)})
exten => _003318364,n,Set(cut1=${CUT(foo,:,2)})
exten
That sounds good, do you have a example of that?
On Fri, Mar 25, 2011 at 9:24 AM, wrote:
> So make a whitelist
>
> What I do is create a outbound route with the allowed cid and then have
> another route which goes to a not allowed recording which catches all other
> caller Id's
> -Original M
On 03/25/2011 08:19 AM, Tilghman Lesher wrote:
On Thursday 24 March 2011 04:50:48 Jonas Kellens wrote:
On 03/24/2011 10:45 AM, Rizwan Hisham wrote:
You have to use adaptive cdr for this functionality. In 1.8 the conf
file for adaptive cdr is cdr_adaptive_odbc.conf. The sample conf fil
One extra line to change "blacklist" to "whitelist"
Exten => _X,1,noop(everybody but 103 dials)
Exten => _X./100,n,Dial(DAHDI/1,w,5551212)
Exten => _X./101,n,Dial(DAHDI/1,w,5551212)
Exten => _X.,n,hangup
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun
On 03/25/2011 04:58 AM, Thomas Winter wrote:
Hi list,
I have an 44100 Hz file with human voice, stereo with 16Bit.
When convertig this to 8 kHz, mono I loose a lot of quality and have
some ground noise. I tried several sox options but without success.
Can somebody help
best regards Thomas
Yo
In 1.4 there was "core show channels concise"
This seems to be gone from 1.8.
When I am using the AMI interface to get a listing of all channels
my listing "names" are cut short.
SIP/devcentos5x64_to
notice above. In 1.4 it would have given me "SIP/devcentos5x64_to_am2mm"
How in 1.8 do I get t
Based on the following URL, it seems that CallWeaver may not still be an active
project??
http://www.callweaver.org/blog/20
>From a security standpoint, I would usually expect it is safer to be with an
>active project, than a dead one. Otherwise who is going to patch
>vulnerabilities? Not me.
Don't have to be a developer to be a patcher, but it helps ...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas
Mortensen
Sent: Friday, March 25, 2011 9:25 AM
To: Asterisk Users Mailing List - Non-Commerc
I have been somewhat interested in FreeSwitch in the past, but I am mostly
interested in Asterisk. That's why I asked about stability of asterisk
versions. Maybe some other time I'll look deeper into FreeSwitch.
Thanks. And thanks everyone for the feedback.
-
Doug Mortensen
Network Consultant
Im
Do you have the same ratio of deployments using 1.4 as you do with 1.2? What
about 1.6 or 1.8? I simply question how accurate a comparison can be made when
one is comparing 50 1.2 deployments to 5 1.4 deployments? I'm sure it says
something, and I do appreciate the feedback.
-
Doug Mortensen
Ne
Hi Doug!
I use Asterisk 1.4 and 1.8, I can easily see that Asterisk 1.8 works better
than 1.4.
Everything on Asterisk 1.8 seems better.
Best regards,
> From: d...@impalanetworks.com
> To: asterisk-users@lists.digium.com
> Date: Fri, 25 Mar 2011 08:32:04 -0600
> Subject: Re: [asterisk-users] Wha
Now that we've hashed out some thoughts on the most stable version of asterisk,
I'd like to ask the question as to why I should NOT use 1.8? What are specific
reasons? For instance a few days back I was speaking with James at Rhino
Equipment. He said that he has "no real data" on why I shouldn't
On Fri, Mar 25, 2011 at 9:51 AM, Jerry Geis wrote:
> In 1.4 there was "core show channels concise"
> This seems to be gone from 1.8.
>
> When I am using the AMI interface to get a listing of all channels
> my listing "names" are cut short.
>
> SIP/devcentos5x64_to
>
> notice above. In 1.4 it would
Ah, makes sense!
Thanks!
On Fri, Mar 25, 2011 at 2:09 PM, Danny Nicholas wrote:
> One extra line to change “blacklist” to “whitelist”
>
> Exten => _X,1,noop(everybody but 103 dials)
>
> Exten => _X./100,n,Dial(DAHDI/1,w,5551212)
>
> Exten => _X./101,n,Dial(DAHDI/1,w,5551212)
>
> Exten => _X.,n
On Fri, 25 Mar 2011, Steve Underwood wrote:
You really need to remove the bass end of the spectrum before downsampling to
8k/s. uLaw/ALaw sound pretty muddy and horrible if you don't do that, and the
other common 8k/s codecs don't sound any better. Jean-Marc Valin wrote a
little filtering util
On Fri, Mar 25, 2011 at 7:36 AM, Douglas Mortensen
wrote:
>
> But I would like specific reasons why I shouldn't use 1.8 in a production
> environment if anyone has some?
That is a loaded question, in that no two environments are likely to
be the same. Some bugs are major issues for < 1% of the
From: "Bob Beers"
Sent: Friday, March 25, 2011 10:44 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: Re: [asterisk-users] asterisk 1.8 question
On Fri, Mar 25, 2011 at 9:51 AM, Jerry Geis wrote:
> In 1.4 there was "core show
From: "Jonathan Thurman"
Sent: Friday, March 25, 2011 11:06 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: Re: [asterisk-users] Why shouldn't I use 1.8?
On Fri, Mar 25, 2011 at 7:36 AM, Douglas Mortensen
wrote:
>
> But I wo
Hi list!
Our company is currently using 3 asterisk boxes in 3 locations
connected through iax2. Our main office makes and receives many more
calls than the other two. I'm looking for a way to check within the
dialplan how many channels are in use at the main office so if it
reaches a threshold out
On Fri, 25 Mar 2011, Nathan Pryor wrote:
Is there a command I could use directly in the dialplan or with the
manager interface to get the number of used channels?
Check out the GROUP() and GROUP_COUNT() functions.
--
Thanks in advance,
-
Great advice guys. I know it was a loaded question. I appreciate your feedback.
Although I'm probably not as much of an asterisk guru as you guys, I tend to
agree with your approach.
Thanks a lot!!
-
Doug Mortensen
Network Consultant
Impala Networks
P: 505.327.7300
.
From: Bryant Zimmerman [mai
On 25/03/11 14:36, Douglas Mortensen wrote:
Now that we've hashed out some thoughts on the most stable version of asterisk, I'd like
to ask the question as to why I should NOT use 1.8? What are specific reasons? For
instance a few days back I was speaking with James at Rhino Equipment. He said
A quick question. When looking at issues.asterisk.org, It allows issues/bugs to
be filtered by "Asterisk Version". The 1.8.x options for the filter are:
1.8.2.3
1.8.2.4
1.8.3.2
1.8.4-rc2
Do you guys know whether bugs from the older version should still show up as
issues in the newer versions as
On Fri, Mar 25, 2011 at 11:36 AM, Steve Edwards
wrote:
> On Fri, 25 Mar 2011, Nathan Pryor wrote:
>
>> Is there a command I could use directly in the dialplan or with the
>> manager interface to get the number of used channels?
>
> Check out the GROUP() and GROUP_COUNT() functions.
>
That's what
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas
Mortensen
Sent: Friday, March 25, 2011 12:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Why shouldn't I us
On 24/03/11 05:49, Olivier CALVANO wrote:
The To, "To:", can i get it into
a variable for sent it at a API ?
You want the sip_header function:
http://www.voip-info.org/wiki/view/Asterisk+func+sip_header
cheers,
Paul.
--
_
-
Has anyone had any luck getting this phone up and running on an asterisk
server, most noticeably a Trixbox installation?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
Hey Guys!
I have two asterisk 1.8.3.2 same version on both machine but why one asterisk
has "reload" command but other doesn't ?
satish-desktop*CLI> core show version
Asterisk 1.8.3.2 built by root @ satish-desktop on a x86_64 running Linux on
2011-03-25 16:10:39 UTC
satish-desktop*CLI> re
re
On 11-03-25 02:49 PM, satish patel wrote:
I have two asterisk 1.8.3.2 same version on both machine but why one asterisk has
"reload" command but other doesn't ?
*CLI> module reload
'reload' is no longer a valid command. I'm guess one box has
cli_aliases.conf, while the other does not.
--
P
Both servers files are identical..
root@satish-desktop:~# cat /etc/asterisk/cli_aliases.conf | grep reload
reload=module reload
; Alias for making voicemail reload actually do module reload app_voicemail.so
;voicemail reload=module reload app_voicemail.so
; This will make the CLI command "mr"
All-
My apologies in advance if this is an obvious question and I've missed it on
Asterisk FAQs and how-to's...
Can Asterisk operate with just an FXO card? By that I mean, no network
connection (none, no local network). I want
to build some type of user interface to go off-hook, route FXO por
Following is my scenario to connect back to back PRI of two asterisk server.
PRI cards are Sangoma A102D
[Asterisk1][PRI]-Cross Cable-[Asterisk2]
Asterisk1
; Span 1 (MASTER)
switchtype = national ; commonly referred to as NI2
context = from-pstn
group = 0,24
echocancel =
On 11-03-25 03:13 PM, satish patel wrote:
Both servers files are identical..
*CLI> module show like cli
--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com & http://asterisk.org
--
satish-desktop*CLI> module show like cli
Module Description Use
Count
res_clioriginate.soCall origination and redirection from th 0
res_clialiases.so CLI Aliases 0
2 modules loaded
shirle
No kidding.. found this line second server. Thanks!!
root@shirley:/# cat /etc/asterisk/modules.conf | grep res_clialiases.so
noload => res_clialiases.so
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Fri, 25 Mar 2011 19:53:58 +
Subject: Re: [asterisk-users] rel
1.4 is the new flavor for my new deployments, but I definitely have
more (way more, like 1:8) 1.2 systems in production.
On Fri, Mar 25, 2011 at 10:32 AM, Douglas Mortensen
wrote:
> Do you have the same ratio of deployments using 1.4 as you do with 1.2? What
> about 1.6 or 1.8? I simply question
Did you check so see if the pri is up?
Also, make sure wanpipe & dahdi is setup correctly.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel
Sent: Friday, March 25, 2011 3:41 PM
To: asterisk-users
Subject: [asterisk-u
On Friday 25 March 2011 04:36:28 Jonas Kellens wrote:
> On 03/25/2011 08:19 AM, Tilghman Lesher wrote:
> > On Thursday 24 March 2011 04:50:48 Jonas Kellens wrote:
> >> On 03/24/2011 10:45 AM, Rizwan Hisham wrote:
> >>> You have to use adaptive cdr for this functionality. In 1.8 the conf
> >>> file
satish patel wrote:
group = 0,24
Granted, I'm still running 1.4.x, but I don't believe the above is valid.
My guess is it should be:
group = 0
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety
On Friday 25 March 2011 15:11:49 Doug Lytle wrote:
> satish patel wrote:
> > group = 0,24
>
> Granted, I'm still running 1.4.x, but I don't believe the above is
> valid.
>
> My guess is it should be:
>
> group = 0
No, that's valid. You can have any of groups 0-63 set on a single
group of chann
Asterisk1
satish-desktop*CLI> dahdi show status
Description Alarms IRQbpviol CRC4 Fra Codi
Options LBO
wanpipe1 card 0 OK 0 0 0 ESF B8ZS
0 db (CSU)/0-133 feet (DSX-1)
wanpipe2 card 1
Thanks Doug,
I tried that also but result is same.
> Date: Fri, 25 Mar 2011 16:11:49 -0400
> From: supp...@drdos.info
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue
>
> satish patel wrote:
> > group = 0,24
>
> Granted, I'm still running
On Friday 25 March 2011 14:40:40 satish patel wrote:
> Following is my scenario to connect back to back PRI of two asterisk
> server. PRI cards are Sangoma A102D
>
> [Asterisk1][PRI]-Cross Cable-[Asterisk2]
>
> Asterisk1
>
> ; Span 1 (MASTER)
> switchtype = national ; commonl
One more thing i would like to tell you i have following wanpipe configuration
at both side
@Asterisk1
root@satish-desktop:~# cat /etc/wanpipe/wanpipe1.conf | grep -i clock
TE_CLOCK= MASTER
TE_REF_CLOCK= 0
@Asterisk2
root@shirley:/# cat /etc/wanpipe/wanpipe2.conf | grep -i clock
TE
Okay! i have changed context at master side
; Span 1: WPT1/0 "wanpipe1 card 0" (MASTER)
switchtype = national ; commonly referred to as NI2
context = from-internal
group = 1,24
echocancel = yes
signalling = pri_net
channel => 1-23
Same error nothing change..
satish-desktop*CLI> core set ver
Only Difference is one side card is ECHO Cancellation supported and other is
non-ECHO cancellation. Is there any issue ?
@Asterisk1
Sangoma A102 (non-ECHW)
@Asterisk2
Sangoma A102D (ECHW)
-Satish
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Fri, 25 Mar 2011 20:4
sometime i am getting following error also. what is this means?
[Mar 25 17:11:52] WARNING[5961]: sig_pri.c:985 pri_find_dchan: Span 1: No
D-channels available! Using Primary channel as D-channel anyway!
From: satish...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Fri, 25 Mar 2011 2
I just start "Pri set debug on span 1" and its showing D-channel is down
satish-desktop*CLI> pri show span
Usage: pri show span
Displays PRI Information on a given PRI span
satish-desktop*CLI> pri show span 1
Primary D-channel: 24
Status: Down, Active
Switchtype: Q.SIG switch
Type: Netw
On Friday 25 March 2011 16:23:27 satish patel wrote:
> I just start "Pri set debug on span 1" and its showing D-channel is
> down
How do you have the underlying T1 signalling set up in
/etc/dahdi/system.conf (on both ends)?
--
Tilghman
--
___
Sorry for the crosspost. This was supposed to be on -users
I know some of you are polycom gurus...
Anyone know how to remove transfer from a polycom 33x phone? We've set
allowtransfer=no, but we would like to remove a polycom soft key as well.
--
___
Can anyone recommend some White Papers or Success Cases that we can use
to ease the mind of a customer that has not heard much about Asterisk? All
they know is Avaya at this point.
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161
Both server has same content in system.conf file.
satish@shirley:~$ cat /etc/dahdi/system.conf
# Global data
loadzone= us
defaultzone = us
span = 1,1,0,esf,b8zs
bchan = 1-23
dchan=24
echocanceller = mg2,1-23
> From: tilgh...@meg.abyt.es
> To: asterisk-users@lists.digium.com
> D
On Fri, Mar 25, 2011 at 6:05 PM, Carlos Chavez wrote:
> Can anyone recommend some White Papers or Success Cases that we can use
> to ease the mind of a customer that has not heard much about Asterisk? All
> they know is Avaya at this point.
>
> --
> Carlos Chavez
> Director de Tecnología
> Te
Check out this https://issues.asterisk.org/view.php?id=17270
> From: tilgh...@meg.abyt.es
> To: asterisk-users@lists.digium.com
> Date: Fri, 25 Mar 2011 17:23:28 -0500
> Subject: Re: [asterisk-users] Back-to-back asterisk PRI issue
>
> On Friday 25 March 2011 16:23:27 satish patel wrote:
> > I j
On Fri, Mar 25, 2011 at 7:05 PM, Carlos Chavez wrote:
> Can anyone recommend some White Papers or Success Cases that we can use
> to ease the mind of a customer that has not heard much about Asterisk? All
> they know is Avaya at this point.
>
> --
> Carlos Chavez
> Director de Tecnología
> Te
Hello,
Occasionally, I get the following warning in my asterisk log,
pbx.c: We were unable to say the number [n], is it too large?
n is two or one digit number, which doesn't look like large to me!
Could anybody please tell more about this warning, like in what scenario I
may have this warning
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