Hi,
Today at 12 Non EDT, Dan York will be with us to talk about the recent
on and off moments of Google Voice SIP URI calling. Like Skype +
Asterisk (or any SIP), Google Voice and SIP compose the other shoe
waiting to drop. We're following this with interest. So GV turned on
SIP URI and then a
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 are using
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
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 da...@debsinc.com wrote:
Just use “Ex-girlfriend”
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 peterdenhar...@gmail.com
Sender:
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 ${EXTEN:0:3}
will only return the first 3 digits entered
Ish
On Wed, 2011-03-23 at 16:27 -0400, Eddie Mikell wrote:
All:
Some of the people who dial into to our system will press the pound key
when entering an extension for the directory key. When waitexten gets
that, I get
On Thu, 2011-03-24 at 21:58 +0100, Thomas Winter wrote:
Hi list,
I have an 44100 Hz file with human voice, stereo with 16Bit.
When convertig this to 8 kHz, mono I loose a lot of quality and have
some ground noise. I tried several sox options but without success.
Can somebody help
best
Hi Olivier,
here is solutions for your situation , ideally you need to talk with
Provider and they can set SIP URI
for given DID numbre , but that can be solved by dial-plan like this.
exten = _003318364,1,Set(foo=${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, isr...@gmail.com 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
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
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
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
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 the
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 -
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
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
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] What
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 ge...@pagestation.com 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
Ah, makes sense!
Thanks!
On Fri, Mar 25, 2011 at 2:09 PM, Danny Nicholas da...@debsinc.com 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 =
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
On Fri, Mar 25, 2011 at 7:36 AM, Douglas Mortensen
d...@impalanetworks.com 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
From: Bob Beers bob.be...@gmail.com
Sent: Friday, March 25, 2011 10:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk 1.8 question
On Fri, Mar 25, 2011 at 9:51 AM,
From: Jonathan Thurman jonat...@thurmantech.com
Sent: Friday, March 25, 2011 11:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Why shouldn't I use 1.8?
On Fri, Mar 25,
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
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
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
On Fri, Mar 25, 2011 at 11:36 AM, Steve Edwards
asterisk@sedwards.com 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()
-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
On 24/03/11 05:49, Olivier CALVANO wrote:
The To, To:sip:081169x...@91.121.xxx.xxx;user=phone, 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
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 tabtab
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.
--
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
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
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]
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
d...@impalanetworks.com wrote:
Do you have the same ratio of deployments using 1.4 as you do with 1.2? What
about 1.6 or
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:
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 for adaptive
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
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 channels.
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 1.4.x,
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 ; commonly
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
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
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
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
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 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:
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:
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
On Fri, Mar 25, 2011 at 6:05 PM, Carlos Chavez cur...@telecomabmex.comwrote:
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
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 just
On Fri, Mar 25, 2011 at 7:05 PM, Carlos Chavez cur...@telecomabmex.com 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
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
64 matches
Mail list logo
