It looks to me that u are having clock synchronism problems due to the fact you
are using Virtual Machine so u don't have an ISDN card generating clock. Are u
using what was called ztdummie as clock source? Can't precise the name of it in
chan_dahdi but u have it.
What u report isn't new and
2010/2/25 Zhang Shukun bit...@gmail.com:
next ,i want to dial from asterisk to PSTN now. i have see the sample
in the extensions.conf relevent to PSTN as follow:
; If you are freely delivering calls to the PSTN, list them here
;
;exten = _1256428,1,Dial(DAHDI/G2/${EXTEN:7}) ; Expose all
Hello list,
I'm having troubles implementing the ${CDR(duration)}
${CDR(billsec)} variables in this scenario:
PEER CALLS OUT -
CALL GOES TO PEER'S DEFAULT OUTGOING CONTEXT -
THE CALL IS SENT TO A MACRO AND GOES IN HANGUP -
THE CALL RETURNS TO EXTENSION h OF
System have been working great for weeks, using an average 40 of 120
dahdi channels.
But today, I suddenly see scary things like this:
-- Moving call from channel 5 to channel 7
[Feb 25 10:18:12] WARNING[17129]: chan_dahdi.c:10608
pri_fixup_principle: Can't fix up channel from 5 to 7 because
Thanks to Tzafrir for the above mentiong documentation.
FYI
http://docs.tzafrir.org.il/dahdi-linux/README.html
A PBX system should generally have a single clock. If you are connected to a
telephony provider via a digital interface (e.g: E1, T1) you should also
typically use the provider's clock
This is a real 'newbie' type question, but I can't get my brain to work
today.
Is it possible to re-direct an incoming SIP call based on it's CLI?
Ideally I would like to check incoming calls against a short whitelist
(of say 20 numbers) and redirect to a different extension if there is a
match.
Hi,
Does anyone have sample configuration files for a Cisco 7905G to use
with SIP/Asterisk ?
I'm on Firmware 3-08-12 - is there a better release to run ?
/S
--
_
-- Bandwidth and Colocation Provided by
On 2/25/10 6:50 AM, Tilghman Lesher wrote:
DHCP is designed in such a way that you can legitimately have multiple DHCP
servers on the same network. The first DHCP server which replies and meets
the DHCP client's requirements will be the server to which the client
registers. If the Linksys
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
Has example
exten = s,1,Answer
exten = s/9184238080,2,Set(CALLERID(name)=EVIL BASTARD)
exten = s,2,Set(CALLERID(name)=Good Person)
exten = s,3,Dial(SIP/goodperson)
for white list
exten =
On 25 Feb 2010, at 02:16, Zhang Shukun wrote:
there is a AudioCodes Mediant 2000 out there. i want to realise ip to
PSTN and PSTN to ip connection.
Ok.
after some configuration on AudioCodes Mediant 2000, PSTN to ip
connecttion works.
Thats good.
a, Do i need install DAHDI or libpri in
On Thursday 25 February 2010 01:29:37 voipas wrote:
Is it possible use asterisk curl function with ssl sertificate?
If you're talking about just connecting to an SSL server, that is dependent
upon using a version of libcurl with SSL support. If, on the other hand,
you're talking about using a
Beginner to Asterisk, but not beginner to VoIP
FreePBX front end running on a dell 1550 and XLite running on a different
Woindows XP box
Both boxes connected via switch on same subnet. No NAT involved
On FreePBX I created a new extension 1001 with a SIP password of 1001
On Xlite, username is
Check your Topology tab - the ICE setting tweaks connectivity if memory
serves (I now have McAfee AV so can't use my Xlite to verify this :( )
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Girard,
Jeffrey COL
On Thu, 2010-02-25 at 03:00 -0800, Kyle Kienapfel wrote:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
Has example
exten = s,1,Answer
exten = s/9184238080,2,Set(CALLERID(name)=EVIL BASTARD)
exten = s,2,Set(CALLERID(name)=Good Person)
exten =
Hi all,
i've 2 asterisk box with dahdi (server A ver. 1.4.29 and server B ver.
1.4.26) connected with IAX channel using gsm codec.
- Calling from A to B the call has no problem: ring , answer a speak
without problem.
- Calling from B to A : B phone always listen ring also when A phone
answer.
Do a core set verbose 10 and repeat the test. CLI should tell you what
you need to handle the exception.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian
Sent: Thursday, February 25, 2010 9:12 AM
To:
Hello list,
when installing Dahdi, the following error comes up :
You do not appear to have the sources for the 2.6.18-164.11.1.el5xen kernel
installed.
make[1]: *** [modules] Error 1
The running kernel version :
-bash-3.2# uname -a
Linux vds.hosting.net 2.6.18-164.11.1.el5xen #1 SMP Wed
Hi to all asterisk-users ;)
As some of you may know, Kernel 2.6.32 includes a module for the
infamous Fritz passive ISDN cards in conjunction with mISDN.
I just would like to know if anybody has tried to use a Fritz card as a
BRI adapter for Asterisk with the new module. If so, i would be
On Thu, Feb 25, 2010 at 9:30 AM, jonas kellens jonas.kell...@telenet.bewrote:
Isn't the kernel the same as the sources ??
Package kernel-devel-2.6.18-164.11.1.el5.x86_64 already installed and latest
versionPackage kernel-headers-2.6.18-164.11.1.el5.x86_64 already installed
and latest
Hello all,
I'm running Asterisk 1.2.35 with chan_mgcp activated.
The process host around 2,4K users.
Along the day I've got some debug reports like :
Feb 24 22:25:42 DEBUG[28546] channel.c: Avoiding deadlock for
'MGCP/aaln/1...@028421223635-1'
Feb 24 22:29:04 DEBUG[28670]
On 23 February 2010 13:16, Razza razz...@gmail.com wrote:
On 23 February 2010 12:58, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
Have you managed to install those zaphfc drivers?
Those are basically the same ones from http://code.google.com/p/zaphfc/
Hi Tzafrir. I checkout out that but
Which Avaya system are you running?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
Sent: Wednesday, February 24, 2010 5:52 PM
To: 'Asterisk Users List'
Subject: Re: [asterisk-users] Which
On 02/25/2010 11:19 AM, Vinícius Fontes wrote:
I'm playing around with an ALIX 2D2 board
(http://www.pcengines.ch/alix2d2.htm). It's a fanless, x86 system using an
AMD Geode processor with 256MB of RAM. Also available are two network
interfaces, two USB ports and one serial port (no
- Shaun Ruffell sruff...@digium.com escreveu:
On 02/25/2010 11:19 AM, Vinícius Fontes wrote:
I'm playing around with an ALIX 2D2 board
(http://www.pcengines.ch/alix2d2.htm). It's a fanless, x86 system
using an AMD Geode processor with 256MB of RAM. Also available are two
network
Hi,
we are running Asterisk 1.6.1.14 and have a issue that when we use followme the
call is correctly placed to the mobile phone, the mobile rings, but when
answered we do not hear the normal followme introduction message. If we press
1 to accept there is just silence. Has anybody else seen
marco.mo...@gmail.com wrote:
It looks to me that u are having clock synchronism problems due to
the fact you are using Virtual Machine so u don't have an ISDN card
generating clock. Are u using what was called ztdummie as clock
source? Can't precise the name of it in chan_dahdi but u have it.
Jeff Brower wrote:
How did you measure the gaps? Using signal or speech analysis
software to display the recording? If you measure number of samples
between the gaps, does it correspond to multiples of RTP packet
payload length (for example, for 8 kHz G711 multiples of 80 samples
between
On Thu, 25 Feb 2010, Vinícius Fontes wrote:
- Shaun Ruffell sruff...@digium.com escreveu:
On 02/25/2010 11:19 AM, Vinícius Fontes wrote:
I'm playing around with an ALIX 2D2 board
(http://www.pcengines.ch/alix2d2.htm). It's a fanless, x86 system
using an AMD Geode processor with 256MB of
Thank you guys for your feedback.
I consider the upgrading to 1.4.29.1.
Does it can definitively prevent me from this kind of freeze ?
Regards,
Adrien .L
De : Miguel Molina [mailto:mmol...@millenium.com.co]
Envoyé : jeudi 25 février 2010 18:21
À : alemo...@legos.fr; Asterisk
Hello Asterisk community,
Today my asterisk server stop working and i had to reboot the server in
order to make it work again, take a look at the error messages in the CLI at
the time of the crash:
[Feb 25 12:44:20] WARNING[6965] chan_sip.c: sip_xmit of 0x920ae80 (len 545)
to 10.4.2.3:5060
Hi Chris,
Morse code is mainly used for HAM radio activity with Asterisk,
connecting radio repeaters area through Internet.
It's often mandatory (local regulations side) to transmit from time to
time and/or at end of traffic period a Morse sequence including the
repeater (or repeater's owner)
This is just curiosity, but I'm wondering why the Morsecode app has remained
part of the trunk for all of these years. Is there any practical use for this
or is it just an homage to the ghosts of telecommunications past? Does
anybody use the Morsecode app for anything interesting? I'm
snip
Does anybody use the Morsecode app for anything interesting? I'm strangely
fascinated by this core piece of Asterisk functionality.
/snip
Duh! How are we going to spread the word about how to take those alien bastards
down if we don't keep morse code around!?!??!
- Gordon Henderson gordon+aster...@drogon.net escreveu:
On Thu, 25 Feb 2010, Vinícius Fontes wrote:
- Shaun Ruffell sruff...@digium.com escreveu:
On 02/25/2010 11:19 AM, Vinícius Fontes wrote:
I'm playing around with an ALIX 2D2 board
(http://www.pcengines.ch/alix2d2.htm).
Hi!
As some of you may know, Kernel 2.6.32 includes a module for the
infamous Fritz passive ISDN cards in conjunction with mISDN.
Haven't tried/seen that, but mISDN ... I avoid it if I can.
I am willing to try this myself, but i am quite reluctant as i have
wasted weeks with Fritz cards
Hi,
I have two asterisk servers with the same version of 1.4.29.1.
The first server named it as MYE1. MYE1 is an incoming server that can
accept incoming calls from PSTN(ZAP E1).
The second server is a pbx functions server and named it as MYPBX(SIP).
The sip.conf of MYE1 likes below:
[MYPBX]
Jonathan-
How did you measure the gaps? Using signal or speech analysis
software to display the recording? If you measure number of samples
between the gaps, does it correspond to multiples of RTP packet
payload length (for example, for 8 kHz G711 multiples of 80 samples
between gaps) ?
On Thu, 25 Feb 2010, Vinícius Fontes wrote:
Just checked and I'm using the high res timer as well:
Feb 25 17:42:32 voyage vmunix: [ 27.028798] dahdi_dummy: Trying to load High
Resolution Timer
Feb 25 17:42:32 voyage vmunix: [ 27.028816] dahdi_dummy: Initialized High
Resolution Timer
Feb
Asterisk Project Security Advisory - AST-2010-003
++
| Product | Asterisk |
Hi Everyone,
I set up my Asterisk 1.4.24 system and everything works well except when I
dial into another service (like conference calling with GoToMeeting) where I
must enter my pin followed by a pound sign. When I do this, it does not
register - BUT if I press the pound sign a second time
The Asterisk Development Team has announced security releases for the following
versions of Asterisk:
* 1.6.0.25
* 1.6.1.17
* 1.6.2.5
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The releases of Asterisk 1.6.0.25, 1.6.1.17, and
Hi,
I am try to configure Asterisk as PBX system with two interfaces as
shown below. One interface pointing to the local subnet with a SIP phone
and another interface pointing to the external ISP SIP Sever.
SJPhone(X.X.141.32)-(Y.Y.47.149)local-intf-|Asterisk|external-
Hello,
We have a situation where a call comes in, users are notified via an external
process (curl request to web service), and we can't answer the call until a
callee can call in and pickup the call.
How can we implement this functionality?
We tried using :
[caller-inbound-leg]
; code to
I'm trying to get my asterisk server to reinvite. I have two asterisk
servers with public IP's. My users (behind NAT) register on one server
(I'll call it server 1), and for some calls they are transfered over
to the other server (server 2), because that server has the E1's.
I want server 1 to be
In 1.2 you can use rtp debug in the CLI
On Thu, Feb 25, 2010 at 8:27 PM, Alejandro Recarey
alexreca...@gmail.com wrote:
I'm trying to get my asterisk server to reinvite. I have two asterisk
servers with public IP's. My users (behind NAT) register on one server
(I'll call it server 1), and for
Charles Wang wrote:
The sip.conf of MYE1 likes below:
[MYPBX]
type=peer
host=mypbx.abc.com http://mypbx.abc.com
nat=no
disallow=all
allow=g729
canreinvite=yes
qualify=no
context=default
insecure=port,invite
Add sendrpid=yes here.
The sip.conf of MYPBX likes below:
[MYE1]
type=peer
Thank you! it's very helpful
2010/2/25 Steve Howes steve-li...@geekinter.net:
On 25 Feb 2010, at 02:16, Zhang Shukun wrote:
there is a AudioCodes Mediant 2000 out there. i want to realise ip to
PSTN and PSTN to ip connection.
Ok.
after some configuration on AudioCodes Mediant 2000, PSTN
hi, all
after my installation of asterisk and adds-on .
when start astrisk, error accours as follow:
[Feb 26 13:18:08] WARNING[16077]: config.c:2025 find_engine: Realtime
mapping for 'sippeers' found to engine 'mysql', but the engine is not
available
what's wrong with me ?
Thanks.
--
Best
On Thu, Feb 25, 2010 at 11:26 PM, Zhang Shukun bit...@gmail.com wrote:
hi, all
after my installation of asterisk and adds-on .
when start astrisk, error accours as follow:
[Feb 26 13:18:08] WARNING[16077]: config.c:2025 find_engine: Realtime
mapping for 'sippeers' found to engine
yes. mysql run ok
the configuration is ok too. i think
is this error shows asterisk can't find mysql database?
2010/2/26 Warren Selby wcse...@selbytech.com:
On Thu, Feb 25, 2010 at 11:26 PM, Zhang Shukun bit...@gmail.com wrote:
hi, all
after my installation of asterisk and adds-on .
when
On Friday 26 February 2010 00:09:55 Warren Selby wrote:
On Thu, Feb 25, 2010 at 11:26 PM, Zhang Shukun bit...@gmail.com wrote:
[Feb 26 13:18:08] WARNING[16077]: config.c:2025 find_engine: Realtime
mapping for 'sippeers' found to engine 'mysql', but the engine is not
available--
Is MySQL
2010/2/26 Tilghman Lesher tles...@digium.com:
On Friday 26 February 2010 00:09:55 Warren Selby wrote:
On Thu, Feb 25, 2010 at 11:26 PM, Zhang Shukun bit...@gmail.com wrote:
[Feb 26 13:18:08] WARNING[16077]: config.c:2025 find_engine: Realtime
mapping for 'sippeers' found to engine 'mysql',
On Thu, Feb 25, 2010 at 8:00 PM, David Gibbons d...@videon-central.com wrote:
Duh! How are we going to spread the word about how to take those alien
bastards down if we don't keep morse code around!?!??!
And what about if you're trapped in ship that sinks? What if the 3g
coverage isn't good?
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