2 nov 2010 kl. 17.19 skrev Olivier:
> Hi,
>
> In Europe many Telcos implement power-save mode
> (See http://www.isdn-test.de/ihhe12.htm#Heading37 and scroll down to
> 'Activation / Deactivation' for more information).
>
> Would you agree to have this feature added to the ones already discuused
31 okt 2010 kl. 13.43 skrev Paul Belanger:
> On Sat, Oct 30, 2010 at 6:22 PM, Brian Capouch wrote:
>> I wonder if anyone out there has a perspective on this. There are a
>> welter of tickets out there on the matter, most of them closed.
>>
> I'm actually able to reproduce this pretty often, fo
6 nov 2010 kl. 15.30 skrev Hans Witvliet:
> Hi all,
>
> As stated in the subject, slightly off-topic, as it is not directly a
> Asterisk issue, but more SIP in general
>
> Because security in general, and specifically identification becomes
> more and more a subject for more concern, and Asteri
10 nov 2010 kl. 02.38 skrev Brett Woollum:
> Good idea Paul.
>
> My debug output:
> [Nov 9 17:33:39] VERBOSE[2923] netsock.c: == Using SIP RTP CoS mark 5
> [Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:1]
> Set("SIP/413-0005", "CALLERID(num)=2") in new stack
Hi All,
I am running asterisk on Linux machine and trying to use confbridge
application. Please have a look at Conf files.
sip.conf
==
[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow = all
allow=ulaw
allow=alaw
defaultexpiry=100
[5
Good idea Paul.
My debug output:
[Nov 9 17:33:39] VERBOSE[2923] netsock.c: == Using SIP RTP CoS mark 5
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipphones:1]
Set("SIP/413-0005", "CALLERID(num)=2") in new stack
[Nov 9 17:33:39] VERBOSE[4175] pbx.c: -- Executing [...@sipph
On Tue, Nov 9, 2010 at 6:35 PM, Brett Woollum wrote:
> Nobody has any idea why the Caller ID is being overwritten when using
> Asterisk Realtime for the SIP users?
>
No, perhaps you can _show_ us the problem.
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
--
Paul Belange
Nobody has any idea why the Caller ID is being overwritten when using Asterisk
Realtime for the SIP users?
Brett Woollum
br...@woollum.com
- Original Message -
From: "Brett Woollum"
To: asterisk-users@lists.digium.com
Sent: Sunday, November 7, 2010 3:08:50 PM GMT -08:00 US/Canad
Hi, How to enable zaptel debugging?
I need to see reverse polarity messages.
Thank you,
Imran
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asteri
This can be done using the directory on the Aastra phone. Not sure what
you have access to with Trixbox Pro. Look in the Aastra documentation
for info on setting up the directory.csv file.
--
_
-- Bandwidth and Colocation Provi
Is there a way running Trixbox Pro and Aastra 6731i phones to display the
name of the extension you are trying to dial?
For example, I want to dial John Smith at x4000, I pick up my phone, dial
x4000 and it displays John Smith?
Thanks
--Dovey
--
_
Un-top-posting...
> 2010/11/5 Mickael MONSIEUR
> Have you noticed a marked increase in CPU load when using MixMonitor?
>
> I use PHPAgi and Asterisk 1.6.2.9-2.
2010/11/5 Norbert Zawodsky
Obviously, if the box has more to do, CPU load will increase.
What do you mea
Not sure, but you can try to increase debug log level and check whether
you'll have more details
On Tue, Nov 9, 2010 at 4:55 PM, Mickael MONSIEUR wrote:
> You think of a loop?
> This is possible because I use AGISIGHUP=no ..
>
> exten => s,1,set(AGISIGHUP=no);
> exten => s,2,AGI(myapp.agi) ;
>
Hi
In Asterisk 1.6/realtime Mysql, we can't put a username/password in a
Dial Command ?:
'Dial', 'SIP/Username:passw...@mypeer/${EXTEN},180,r'
Thanks
Olivier
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.
On Tue, Nov 9, 2010 at 2:09 AM, Tilghman Lesher wrote:
> On Monday 08 November 2010 16:05:28 Carlos Chavez wrote:
>> On Mon, 2010-11-08 at 16:53 -0500, bakko wrote:
>> > The addons are in the same package.
>> >
>> > Regards
>> > - Original Message -
>> > From: "Carlos Chavez"
>> > To: "As
Try kannel http://www.kannel.org
It' a very good and powerful WAP and SMS gateway.
Adolphe Cher-aime
From my Iphone
On Nov 9, 2010, at 10:35 AM, Flavio Miranda
wrote:
Hi list,
Anyone has some guidance in how can I project a SMS gateway with
Asterisk. I mean, some good web link,pdf
--[ UxBoD ]-- wrote:
> Has anyone managed to successfully connect Asterisk to Zimbra using the
> Jabber service
>
I did a couple months ago, using GaJim, but haven't been able to
reproduce it. I've since moved on to OpenFire for my Jabber server
I will be revisiting this again, hopefully b
Has anyone managed to successfully connect Asterisk to Zimbra using the Jabber
service ? I have opened http://issues.asterisk.org/view.php?id=18198 as it
keeps failing for me. Am wondering whether it is due to using a self signed
cert.
--
Thanks, Phil
--
_
You think of a loop?
This is possible because I use AGISIGHUP=no ..
exten => s,1,set(AGISIGHUP=no);
exten => s,2,AGI(myapp.agi) ;
I will put lines and debug log file ... I do not think that Asterisk archive
errors AGI script?
2010/11/9 Marino Punturieri
> So it seems not related to MixMonito
On Tue, 9 Nov 2010, Bruce B wrote:
> Thanks for input. Great info. Good to know all this about the router. I see
> you use a 256MB CF card there. Do you use a USB key stick for storage?
No. Things that stick out of boxes in small offices get broken off. (ie.
the type of places that do not have a
On Mon, Nov 08, 2010 at 02:44:26PM -0500, Jeff LaCoursiere wrote:
> >It could be the echo canceller, I had this kind of problem with OSLEC. I
> >also thought the PRI provider was sending clipped audio. I switched to
> >the VPM450 daughterboard and since audio has been crystal clear. What is
> >your
Hi list,
Anyone has some guidance in how can I project a SMS gateway with Asterisk. I
mean, some good web link,pdf or something like that?
Thanks in advanced!!Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormirandaru
--
_
Jonas Kellens wrote:
> On 11/09/2010 02:12 PM, Gareth Blades wrote:
>> Jonas Kellens wrote:
>>
>>> On 11/08/2010 09:50 PM, Jonas Kellens wrote:
>>>
Hello,
SIP DNS SRV records are not working.
My Grandstream uses the SRV records to find the first Asterisk server
>>
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phuong Hoang
Sent: Monday, November 08, 2010 8:31 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Store CDR (call detail record) to Oracle database
Hi all,
Now
It is 1.6.2.13
ABEJIDE, Ayodele A. (CCNA)
+2348039269311
From: asannu...@gmail.com
To: asterisk-users@lists.digium.com
Date: Tue, 9 Nov 2010 07:38:44 -0500
Subject: Re: [asterisk-users] Festival
Hi,
wich version of Asterisk?
If is 1.6.2.13, there is a open issue becouse not
wor
Hi,
After disabling MixMonitor, I realize that my CPU saturates as always!
What my script PHP-AGI is fairly simple!
- I answer a call
- Some menus
- I send the call to another line $this->exec_dial (SIP/provider/NUMBER,
...)
And I was 75-80% using an e4...@2.40ghz! It is not logic !
Please help
Thanks for input. Great info. Good to know all this about the router. I see
you use a 256MB CF card there. Do you use a USB key stick for storage?
Thanks,
Bruce
On Tue, Nov 9, 2010 at 4:09 AM, Gordon Henderson
> wrote:
> On Mon, 8 Nov 2010, Bruce B wrote:
>
> > Yes, it is a small office. I am f
Jonas Kellens wrote:
> On 11/08/2010 09:50 PM, Jonas Kellens wrote:
>> Hello,
>>
>> SIP DNS SRV records are not working.
>>
>> My Grandstream uses the SRV records to find the first Asterisk server
>> to register to. This works.
>>
>> But when I shut down the Asterisk proces on server 1 and I resta
So it seems not related to MixMonitor.
Are you 100% sure that your PHP-AGi script is not looping somewhere?
You should try to understand which is the process that is taken you CPU.
On Tue, Nov 9, 2010 at 2:32 PM, Mickael MONSIEUR wrote:
> Hi,
> After disabling MixMonitor, I realize that my CPU
On 11/09/2010 02:12 PM, Gareth Blades wrote:
> Jonas Kellens wrote:
>
>> On 11/08/2010 09:50 PM, Jonas Kellens wrote:
>>
>>> Hello,
>>>
>>> SIP DNS SRV records are not working.
>>>
>>> My Grandstream uses the SRV records to find the first Asterisk server
>>> to register to. This works.
>>
Dear Asterisk-Users,
I installed festival and while trying to connect it to asterisk it comes up
with:
serverMon Nov 8 18:38:51 2010 : Festival server started on port
1314client(1) Mon Nov 8 18:38:51 2010 : accepted from
localhost.localdomainclient(1) Mon Nov 8 18:38:51 2010 : disconne
Hi,
wich version of Asterisk?
If is 1.6.2.13, there is a open issue becouse not work
https://issues.asterisk.org/view.php?id=17995
R.--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? J
Hello
I'm about to set up a voicemail system for multiple wholesale customers.
So I use a realtime mysql config for the mailboxes.
All single mailboxes have their information about the number, emailaddress,
password in the database. This works fine.
Now the notification emails of course should
On Mon, 8 Nov 2010, Bruce B wrote:
> Yes, it is a small office. I am familiar with pfSense. I am not sure if
> firewall on Astlinux is as versatile and flexible. But also, I am wondering
> if with all those attacks around now-a-days if the box will be able to
> handle 5 extensions, voicemail, IVR,
On 11/08/2010 09:50 PM, Jonas Kellens wrote:
Hello,
SIP DNS SRV records are not working.
My Grandstream uses the SRV records to find the first Asterisk server
to register to. This works.
But when I shut down the Asterisk proces on server 1 and I restart my
GXP 2010, the phone does not regis
35 matches
Mail list logo
