Ok, the problem solved...
thanks for your advice.
after rebooting i run /usr/share/dahdi/xpp_fxloader load
and everything run normally.
thanks...
> Message: 24
> Date: Mon, 1 Feb 2010 11:03:44 +0200
> From: Tzafrir Cohen
> Subject: Re: [asterisk-users] Astribank problem
> To: asterisk-users@l
Using 1.6.2.1 with a TDM400, attached to internal analog phones and
PSTN. When I dial out to PSTN, I cannot send tones, like push "1" for
something stupid. The call itself works, but the DTMF tones fail.
-- Starting simple switch on 'DAHDI/1-1'
-- Executing [6258...@internal:1] Answer("DAHD
David Gibbons wrote:
>
> I have upgraded the phones to the most recent firmware (POS3-08-11-00)
> and * is "Version: 1:1.4.21.2~dfsg-3+lenny1" (debian).
>
>
> That doesn't look like cisco firmware to me... Unless I'm mistaken. What
> version are the phones on? (Settings => Status => Firmware Ve
It appears I am having this same problem with softphones, too. It seems that
if I end my request by pressing # twice(slowly) as opposed to once as
requested by the system, it works.
From: John Regal [mailto:jre...@gmail.com]
Sent: Wednesday, February 03, 2010 5:25 PM
To: 'Asterisk Users Mailin
Hi,
I just deployed new Aastra 9480i phones and when I attempt enter digits on
other systems, like host pin in a GoToMeeting, the servers on the other end
do not get my entries. I am assuming this is a DTMF issue but do not see
anything in this phones config other than turning on the display of th
There's some one who can help me?
I'm using Asterisknow with FreePBX and a Patton 4554 with 2 BRI ports on
2 ISDN lines.
I would like routing the call entering by first BRI to one trunk and
call from second BRI to another trunk.
I have created 2 trunks both registering to Patton with different
i
I have upgraded the phones to the most recent firmware (POS3-08-11-00)
and * is "Version: 1:1.4.21.2~dfsg-3+lenny1" (debian).
That doesn't look like cisco firmware to me... Unless I'm mistaken. What
version are the phones on? (Settings => Status => Firmware Versions)
-Dave
--
___
OK! you're right Mr William, that worked, unfortunately i don't know if
there is any points to affect to somebody who helped you (like rating in
other forums) :(, but one more request can you recommand me a book to master
and tweak dialplans?
2010/1/29 William Stillwell (Lists)
> You are using
Hi,
Our Cisco 7940 phones on a single network sometimes seem to drop calls
as soon as they are picked up. After a second INVITE the phone sends
'500 Internal Error'. One phone thinks its still in a call (though there
is no audio) while the other phone is not in a call.
The drop happens immedia
Thanks! That was easy. :)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes
Sent: Wednesday, February 03, 2010 3:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
On 3 Feb 2010, at 19:17, John Regal wrote:
> Can anyone tell me how I could originate a call from my server? My
> use case is I am on the road and I want to dial into the server
> using my cell phone, log into my user account and then enter a
> number to dial so that the call comes from the s
- "Ira" wrote:
> Just Asterisk 1.6.2 on CENTOS 5. I can't imagine any problems at all,
>
> I don't think top has ever been over a couple percent except when I'm
>
> re-building asterisk.
>
> It's got 2Gb of ram so everything is in ram, I've never seen the swap
>
> file usage get past zero
Hi All,
Can anyone tell me how I could originate a call from my server? My use case
is I am on the road and I want to dial into the server using my cell phone,
log into my user account and then enter a number to dial so that the call
comes from the server.
Thanks,
John
--
_
Just Asterisk 1.6.2 on CENTOS 5. I can't imagine any problems at all,
I don't think top has ever been over a couple percent except when I'm
re-building asterisk.
It's got 2Gb of ram so everything is in ram, I've never seen the swap
file usage get past zero. Right now there's 512K free so it's a
The Atom boards are fine for Asterisk and even some "clunkyness"...
Transcoding is where the Atom boards could cause issues but this is in
the realm of 50 channels or so...
~
Andrew "lathama" Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Lear
Ira,
Are you running anything clunky on there like FreePBX or mysql? And
based on your experience with the Atom board, do you think you would
have performance problems with said clunkiness?
Thanks,
Lyle
On Wed, 2010-02-03 at 09:50 -0800, Ira wrote:
> At 11:19 PM 2/2/2010, you wrote:
> >Has anyon
At 11:19 PM 2/2/2010, you wrote:
>Has anyone tried running Asterisk + CentOS 5 on this (or any other)
>Atom board? Is the Atom platform able to handle the load of all the
>interrupts a TE110P or TDM400P card will generate ?
I run 3 pots lines into a TDM04 on an Atom 330 board with no
problems. 3
On Wednesday 03 February 2010 09:24:34 jonas kellens wrote:
> I would like to set the CDR(src)-variable to the SIPphone that is
> initiating the call.
>
> When calling out, the src-variable is always the public telephone
> number.
>
> I get the ERROR : ast_cdr_setvar: Attempt to set the 'src' read-
Richard Kenner wrote:
> Is there a version of the Asterisk core sounds in English done by June
> Wallack? Some folks here prefer her voice to Allison's, but we'd like
> consistency. And is there a version of the Cepstral software with her
> voice?
No, neither of those exist, and I'm not aware of
Hello list.
I would like to set the CDR(src)-variable to the SIPphone that is
initiating the call.
When calling out, the src-variable is always the public telephone
number.
I get the ERROR : ast_cdr_setvar: Attempt to set the 'src' read-only
variable!
Is there some way to implement this ??
Kin
I've tried that as well prior to sending the initial email with no
results.
I'll play some with DISA today.
James Shigley
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Tuesday, February 02, 2010 2:50 PM
T
Is there a version of the Asterisk core sounds in English done by June
Wallack? Some folks here prefer her voice to Allison's, but we'd like
consistency. And is there a version of the Cepstral software with her
voice?
--
_
-- B
Hi!
I've been on this list for over 3 years and this is my first post.
We have a reporting application for Asterisk that is soon to be in beta.
It's a windows application that generates reports from log files (CDR and
queue).
It has a drag and drop approach to report creating. There is a pivot g
Hardware:
Digium TE110P REV.C and REV.D
Gigabyte GA-965G-DS3 Bios F8b
cat /proc/cpuinfo
model name : Intel(R) Core(TM)2 CPU 6600 @ 2.40GHz
stepping: 6
cpu MHz : 2400.080
cache size : 4096 KB
...
latest libpri, dahdi, asterisk as of tonight.
linu
On Wed, Feb 3, 2010 at 12:17 AM, Ben Dinnerville wrote:
>
> This is usually due to an error with the SIP stack not being loaded due
> to an error - make sure that full logging is on and check your log file
> and search for ERROR and see if there is any mention to SIP (chan_sip.o
> etc), alternativ
3 feb 2010 kl. 08.11 skrev Alex Balashov:
> On 02/03/2010 02:03 AM, Olle E. Johansson wrote:
>>
>> 2 feb 2010 kl. 11.20 skrev BERGANZ Francois:
>>
>>> Hello all,
>>>
>>> Does asterisk accept uri tel: instead of sip: ?
>>>
>>>
>> No, but I think it would be a good addition.
>
> Why? Just cu
On Wed, 3 Feb 2010, Remco Barendse wrote:
> I currently have some Asterisk home servers on general pc hardware as well
> as a mission critical server asterisk pbx running on a Dell 2850
>
> To reduce noise and power consumption i would like to migrate them all to
> an Intel Atom based solution, sh
OK. thanks for the replies. I missed the point about expiration, i will focus
on this point.
On Wed, Feb 03, 2010 at 08:58:00AM +0200, Mindaugas Kezys wrote:
> Just remember, that after reload you will lose all registrations.
>
> Regards,
> Mindaugas Kezys
> http://www.kolmisoft.com
> VoIP Billi
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