4:05 PM
Subject: Re: [asterisk-users] only ring phones that are not on a call
> On Tue, 17 Feb 2009, Jon Weisman wrote:
>
>> is there anything i can do in my dialplan to only ring phones which are
>> not
>> on a call at the time someone dials in?
>>
>> its for a c
is there anything i can do in my dialplan to only ring phones which are not
on a call at the time someone dials in?
its for a call center, they do not want to use queues, but they are
complaining that the call waiting beep is annoying.
i tried call-limit in the sip.conf but then it just busy ou
since you're using ulaw
try setting dtmfmode = inband
if this doesnt work try = auto
-Jon
- Original Message -
From: "Christopher Gray"
To: "Asterisk Users Listserve"
Sent: Friday, January 23, 2009 8:13 PM
Subject: [asterisk-users] Passing DTMF
> Hello:
>
> I need to be able to re
ok what about people that have no choice but to use MS CRM?
- Original Message -
From: "Louis-David Mitterrand"
To:
Sent: Wednesday, January 21, 2009 8:04 AM
Subject: Re: [asterisk-users] integration with Microsoft CRM?
> On Wed, Jan 21, 2009 at 12:58:51PM -, Andrew Thomas wrot
asterisk does pass thru out of the box, there is nothing to install.
in your sip.conf
just add the following:
disallow=all
allow=g729
this will force the peer to use g729 and the end points will take care of the
codec assuming both end points support g729 to begin with.
-jon
- Origina
awesome thanks!
- Original Message -
From: "Kevin P. Fleming"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, January 15, 2009 9:48 AM
Subject: Re: [asterisk-users] G729 host id
> Jon Weisman wrote:
>> So i made a bac
So i made a backup long time ago of the g729 license file for one of my
servers, problem is I dont remember which one. Anybody know how I can
identify which server this license file belongs to?
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Could you further clarify on this? Why is the norm shifting from 9 to 8?
-Jon
- Original Message -
From: Lyle Giese
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, January 09, 2009 8:45 AM
Subject: Re: [asterisk-users] Not Dialing 9
Gordon Hender
Just moved over to DAHDI today and have a few questions.
My environment is:
Asterisk 1.4.22
Using a TE410P
4 PRI's
1. Now when calls come in on those PRI's I get this message in the console:
[Dec 6 19:42:18] WARNING[31557]: chan_dahdi.c:1481 dahdi_enable_ec: Unable to
enable echo cancellation
ers] RTCP too short
> Do you have Grandstream phones?
>
> I noticed a similar issue last year with Grandstream GXP2000 phones. The
> phone was sending an empty RTP packet for the "keepalive" whilst on
> mute. I reported a bug to Grandstream but nothing happened.
>
> r
too short
Double check your config files. Rtp.c is a real-time component, so you're
getting a "phantom call" to this routine (possibly from CDR?)
--
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Weisman
Sent: Friday, November 28, 2008 11:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RTCP too short
I get this all the time. Still haven't found a
I get this all the time. Still haven't found a solution but it doesnt seem to
affect call quality or server performance. I think there's a way to disable the
message, but I lost the link. :(
-Jon
- Original Message -
From: michel freiha
To: Asterisk Users Mailing List - Non-Comme
the span is KO, you can use any applications to have to send you a
alarm email.
Giorgio Ciccarelli
Jon Weisman wrote:
Thanks Hakan,
I was kind of hoping I wouldn't have to "write" anything. Anybody else got
something I could just use?
- Original Mess
is this for php?
- Original Message -
From: "federico fetto" <[EMAIL PROTECTED]>
To:
Sent: Wednesday, November 19, 2008 8:41 AM
Subject: Re: [asterisk-users] Monitoring
> On Wed, 19 Nov 2008 15:17:50 +0200
> "Hakan C" <[EMAIL PROTECTED]> wrote:
>
>> Hey Jon,
>>
>> You are asking so
f you want to monitor your PRI, its not so difficult to script.
See?
It doesnt need write something huge.
Hope it helps.
Thanks.
On Wed, Nov 19, 2008 at 2:29 PM, Jon Weisman <[EMAIL PROTECTED]> wrote:
Thanks Hakan,
I was kind of hoping I wouldn't have to "wri
7;, it returns PRI status.
Good lucks
On Wed, Nov 19, 2008 at 1:57 PM, Jon Weisman <[EMAIL PROTECTED]> wrote:
Hello all -
We are trying to implement some monitoring systems for our production
asterisk boxes. We use whats up gold for all our other stuff. I'd like
Hello all -
We are trying to implement some monitoring systems for our production
asterisk boxes. We use whats up gold for all our other stuff. I'd like to be
able to monitor the status of PRI's. For example if a PRI is in alarm, i'd
like to get an e-mail notification. How are others accomplish
Hello,
Today I saw about 40 calls drop on my asterisk box. Its doing Zap to SIP w/
g729 compression. Wasnt sure what the problem is and now I'm monitoring the
console and I see these strange errors. I'm running Asterisk 1.2.24
Nov 6 05:09:09 WARNING[2581]: chan_zap.c:7919 pri_fixup_principle:
All,
I'm planning on getting a Dell PowerEdge 1950. We want to use our Digium
TE410P card, calls will come in TDM and go out VoIP, we will require to
compress them using G729. What specs do I need to support for 4 E-1's with
cdr logging to mysql? We're thinking about getting two servers 4 E-1's
> I do, however, suggest supplying a timeout argument to your Dial()s.
>
> -- Alex
>
> Jon Weisman wrote:
>
>> I'd like to do the following can someone guide me on how to accomplish
>> this?
>>
>>
>> Call comes in via PRI and tries to go out
I'd like to do the following can someone guide me on how to accomplish this?
Call comes in via PRI and tries to go out via SIP if for some reason the ISP
is down and the call can not go out i want it to fail over and send the same
call through a different PRI.
I was thinking something like thi
[Aug 20 10:31:17] WARNING[13825]: rtp.c:892 ast_rtcp_read: RTCP Read too
short
I've got this message scrolling like crazy on my console when I have calls
up. The calls are from TDM to SIP. Did a google search, but wasn't able to
find anything that made sense to me. Any thoughts?
Thanks,
Jon
Hey guys its me again! So I need to setup our Asterisk server with multiple
IP providers. The server has two NICs, we have two providers, now we want
redundancy! Any guides on how to set this up? We're running Fedora Core 5 w/
Asterisk 1.4.
What I would like is that incase one link goes down, o
new
> how "compatible" Asterisk can be.
>
> Anyways, don't forget to sing praises to everyone that was of help and
> CC their bosses.
>
> Thanks,
> Steve Totaro
>
> On Thu, Aug 14, 2008 at 11:00 AM, Jon Weisman <[EMAIL PROTECTED]> wrote:
>> Thanks
t; <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, August 13, 2008 10:22 AM
Subject: Re: [asterisk-users] Sending Set Asynchronous Balanced Mode
Extended
> On Wed, Aug 13, 2008 at 9:56 AM, Jon Weisman <[EMAIL PROTECTED]&g
So we've got a TE410P configured as E-1. The PRI is showing up as normal, I
have green lights, but d channel doesnt seem to come up and i keep getting this
error if i do a "pri intense debug"
The carrier swears up and down that there are no issues on their end. Any
thoughts?
localhost*CLI>
>
Larry,
Give us a call (646) 862-1555
/jon
- Original Message -
From: Larry Costigan
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, July 15, 2008 2:22 PM
Subject: [asterisk-users] Toll Free International Number
Hello All,
I am looking to find
on Kelly
> PCF Corp
> Real Support for your Virtual Office TM
> 651 842-1000
> 888 Don Kell(y)
> 651 842-1001 fax
>
>
>
> -----Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Jon Weisman
> Sent: Wednesday, July 02, 2008 12:
Guys,
Would there be any interest in $1 per DID's/month (U.S.-48 Nationwide) with
a per minute charge of $0.01 on the inbound? I may have something available
if there is enough people interested. Please e-mail me off-list.
/jon
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Anybody here have experience they could share on this switch?
Thanks,
Jon
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Anybody know how to get the tmobile blackberry 8320 to connect to asterisk
via voip? The phone is wifi enabled and does use voip for tmobile.
TIA,
Jon
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asterisk-users mailing l
Steven,
Could you provide more information regarding the tmobile adapater you used?
A link perhapse?
Thanks,
Jon
- Original Message -
From: "Steven" <[EMAIL PROTECTED]>
To:
Sent: Monday, November 05, 2007 7:40 AM
Subject: Re: [asterisk-users] Need Reference sites
> We have implement
Friday, November 09, 2007 12:06 AM
Subject: Re: [asterisk-users] If caller id is null set to a specific number
> Doug Lytle wrote:
>> Jon Weisman wrote:
>>> All,
>>>
>>> If someone calls into my asterisk box and has a private number I would
>>> like to set th
l set to a specific number
> Jon Weisman wrote:
>> All,
>>
>> If someone calls into my asterisk box and has a private number I would
>> like to set the callers id to a specific telephone number, only when
>> the ANI is missing, otherwise if present just pass i
Dan,
What happen w/ this? Did you figure it out? I've setup w/ XO using ESF/B8ZS,
and 5ESS for the switchtype, worked great and got the ANI as well. I dont
think you can get ANI on E&M Wink trunks, how about feature group d?
-Jon
- Original Message -
From: "Dan Casey" <[EMAIL PROTECT
All,
If someone calls into my asterisk box and has a private number I would like to
set the callers id to a specific telephone number, only when the ANI is
missing, otherwise if present just pass it along. Any ideas?
TIA,
Jon___
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;
Sent: Tuesday, October 16, 2007 3:09 PM
Subject: Re: [asterisk-users] tech prefix
Jon Weisman wrote:
How can I add a prefix to an outbound call?
_X. => {
Dial(tech/123{EXTEN});
}
?
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
no that didnt work.
- Original Message -
From: "Philipp Kempgen" <[EMAIL PROTECTED]>
To: "Asterisk Users"
Sent: Tuesday, October 16, 2007 3:09 PM
Subject: Re: [asterisk-users] tech prefix
Jon Weisman wrote:
How can I add a prefix to an outbound call?
_X
There used to be a prefix command in asterisk, it doesnt seem to work any more.
How can I add a prefix to an outbound call?
Thanks,
Jon
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To UNSUBSCRIB
y to require to
> do Answer after each Dial and then send to voicemail.
>
> Regards,
> Atis
>
>>
>> Von: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] Im Auftrag
>> von Jon Weisman
>
>> I've got a strange issue here. When I make a SIP call to
Doug,
Not sure on the trixbox side but for asterisk:
Asterisk server: 10.0.0.1
Cisco Gateway: 10.0.0.2
In sip.conf
[cisco]
context=cisco
type=friend
host=10.0.0.2
dtmf=rfc2833
extension.conf
exten=>_011.,1,Dial(SIP/[EMAIL PROTECTED])
In the Cisco:
dial-peer voice 100 voip
application session
All,
I've got a strange issue here. When I make a SIP call to say my voicemail app,
I hear audio just fine. However when I dial from PSTN into my Asterisk box, I
see that its playing the voice files, but I hear nothing, then the call drops.
I'm running Fedora Core 6, and Asterisk 1.2.24. CLI ou
nc Errors
Slip Events
1 Error every second
Zaptel.conf
loadzone= us
defaultzone = us
span=1,1,0,esf,b8zs
span=2,2,0,esf,b8zs
span=3,3,0,esf,b8zs
span=4,4,0,esf,b8zs
bchan=1-23
dchan=24
bchan=25-47
dchan=48
bchan=49-71
dchan=72
bchan=73-95
dchan=96
TIA,
Jon Weisman | Sales Engine
Whats wrong w/ voip-info.org?
Jon Weisman | Sales Engineer
International Bell Communications
www.ibell.net
- Original Message -
From: Alex Robar
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, June 26, 2007 8:38 AM
Subject: Re: [asterisk-users
inband is for G711 (uLaw) only.
Try rfc2833
Jon Weisman | Sales Engineer
International Bell Communications
www.ibell.net
- Original Message -
From: "Matthew Fredrickson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sen
Thanks everyone for your help. The agi script works as well as RAND (with
the latest trunk version of asterisk).
-Jon
- Original Message -
From: "Steve Murphy" <[EMAIL PROTECTED]>
To:
Sent: Saturday, October 14, 2006 9:03 PM
Subject: [asterisk-users] Re: Generate Random Numbers in d
Steve,
Is RAND available in the latest trunk or do I need the 1.4 beta?
If I do show function RAND it says its not available.
Thanks,
Jon
- Original Message -
From: "Steve Murphy" <[EMAIL PROTECTED]>
To:
Sent: Saturday, October 14, 2006 12:30 AM
Subject: [asterisk-users] Re: Generat
keeps returning 0
- Original Message -
From: "Alexander Lopez" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, October 13, 2006 11:14 PM
Subject: RE: [asterisk-users] Generate Random Numbers in dialplan
You may need to wrap it in an
http://www.pbxfreeware.org/app_backticks.c
On Fri, 13 Oct 2006, Jon Weisman wrote:
Never mind...it does work!
But how do I add its output to a variable?
Thanks,
Jon
- Original Message - From: "Jon Weisman" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Com
Never mind...it does work!
But how do I add its output to a variable?
Thanks,
Jon
- Original Message -
From: "Jon Weisman" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, October 13, 2006 8:29 PM
Subject: Re:
learn something new every day :)
FYI, $RANDOM returns a value between 0 and 32767.
On Fri, 13 Oct 2006, Alexander Lopez wrote:
System(echo $RANDOM)
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jon
Weisman
Sent: Friday, October 13, 2006 1
Hi All,
Anyone know how to generate random numbers in the
dial plan? I've tried using the RAND function but it doesnt work. Basically I
need to generate a random 5 digit number everytime a particular extension is
dialed and then save that into a variable. Any ideas?
Thanks,
Jon
__
We've got a white route w/ VSNL. $0.09 / min, billing is 1/1 prepaid only.
SIP or H323 w/ G729 Codec. E-mail me off-list for testing.
Thanks,
Jon
- Original Message -
From: "Jerry Romney" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; ;
;
Sent: Wednesday, June 28, 2006 1:44 PM
Subjec
Your restriction is what the service provider allows. Most (that I've
used)
allow g729. I know it uses more bandwidth than g723 but nothing like G711
(ulaw or alaw) and from my experience, the quality is quite reasonable.
uh...g729 uses the least bandwidth
-jon
___
April 2006 16:42, Jon Weisman wrote:
And in Extensions.conf
exten=>_X.,1,Prefix(${ACCOUNTCODE})
exten=>_X.,2,Dial,Zap/g1/${EXTEN}
That won't work for this case, as he needs to enter the access code
*after*
dialing. Right offhand, I can't think of doing anything other than
exec
Gary,
What I do is the following:
In SIP.conf
Add the line "accountcode=" and set it equal to each users unique four digit
pin
example:
[user1]
secret=
accountcode=1234
type=friend
host=dynamic
context=default
canreinvite=no
nat=yes
qualify=2000
disallow=all
allow=g729
And in Extensions.c
lle E Johansson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, April 05, 2006 3:37 PM
Subject: Re: [Asterisk-Users] SIP T
5 apr 2006 kl. 16.40 skrev Jon Weisman:
Anyone know how I can get SIP T working w/ Asterisk?
Anyone know how I can get SIP T working w/ Asterisk?
TIA,
Jon
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Hate to reply to my own post, but figured it out.
Just have to setup the IP Phone to DND.
Thanks,
Jon
- Original Message -
From:
Jon Weisman
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, March 14, 2006 2:36
PM
Subject: [Asterisk
All,
How do I get Asterisk to return a 486 SIP response
intentionally?
Thanks,
Jon
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