I just plug the junper in NT mode with no success.
VoipCrazy
2009/8/15 Paul Hales pdha...@optusnet.com.au:
Use a standard network cable - but you have to activate the 'terminate'
jumper on the NT end.
- Also, the new BRI stuff in dahdi is much easier to work with than misdn.
PaulH
voip
Please Use DIALSTATUS Application variable
exten = s,n,Dial(${ARG1},${ARG2},${ARG3},${ARG4})
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-ANSWER,1,Noop(please do action in next priority)
exten = s-ANSWER,2,Playback(demo-instruct)
exten = s-CANCEL,1,Hangup
exten =
Thanks. DIALSTATUS works except ANSWER. When the phone hangup, the
dialplan doesn't go to s-ANSWER.
-- Executing [3...@default:12] Dial(SIP/10.31.0.32-09872150,
SIP/3001|50|Tt) in new stack
-- Called 3001
-- SIP/3001-0986d1d8 is ringing
-- SIP/3001-0986d1d8 answered
Lee, John (Sydney) wrote:
Thanks Tilghman.
I learnt it the hard way - I never imagined I need to jot down the
serial number of a PCI card :-(
I've had a linecard that's been unregistered now for 4 years or more,
because it's in a production server.
It does of course mean that I didn't get
2009/8/14 Pascal Bruno tipas...@gmail.com
Did you get CDRTool to work with Asterisk or Areski's CDR Stats?
Hi finally use Areski but until now I dont try all features, just CDR Report
button. But I had a quick look on other button and it seems work.
can you send your dialplan
it should work
On Mon, Aug 17, 2009 at 2:39 PM, Rilawich Ango maillist...@gmail.comwrote:
Thanks. DIALSTATUS works except ANSWER. When the phone hangup, the
dialplan doesn't go to s-ANSWER.
-- Executing [3...@default:12] Dial(SIP/10.31.0.32-09872150,
On Monday, August 17, 2009, Rilawich Ango wrote:
Thanks. DIALSTATUS works except ANSWER. When the phone hangup, the
dialplan doesn't go to s-ANSWER.
-- Executing [3...@default:12] Dial(SIP/10.31.0.32-09872150,
SIP/3001|50|Tt) in new stack
-- Called 3001
-- SIP/3001-0986d1d8 is
Hi,
I am trying to install asterisk-1.2.34 but facing following issue. I have
gone through it and found that there are files in /usr/lib
libpq.a libpq.so libpq.so.4libpq.so.4.1
make[1]: Entering directory `/usr/src/asterisk-1.2.34/cdr'
gcc -shared -Xlinker -x -o cdr_pgsql.so
On Mon, Aug 17, 2009 at 06:21:45PM +0600, ast guy wrote:
Hi,
I am trying to install asterisk-1.2.34 but facing following issue. I have
gone through it and found that there are files in /usr/lib
libpq.a libpq.so libpq.so.4libpq.so.4.1
make[1]: Entering directory
The error on system crash is:
Digum Board: TDM2400P
OS: Debian Lenny 5.02
dahdi_ec_chunk 0x3c/0x200 [dahdi] SS:ESP 00068:F747FE38
KERNEL PANIC NOT SYNCING
Luis Morales wrote:
The error on system crash is:
Digum Board: TDM2400P
OS: Debian Lenny 5.02
dahdi_ec_chunk 0x3c/0x200 [dahdi] SS:ESP 00068:F747FE38
KERNEL PANIC NOT SYNCING
Hi everybody,
I have a 1.6.0.5 asterisk system and I’m having many problems with my pstn
cards. This system has a Digium B410P with 4 BRIs configured, and an OpenVox
A400E.
The issue is that it’s impossible to make the system recognize the A400E,
despite the “lspci” output is:
0a:00.0
On Mon, Aug 17, 2009 at 10:10 AM, Joan Antoni Terrenebh...@gmail.com wrote:
Regarding the “wctdm-“ in dahdi_hardware output, I’ve found the following
error at “dmesg”:
wctdm: probe of :0a:00.0 failed with error -5
This A400E is a PCI Express card and I’ve tried to plug it in all 3 PCI
Easy questions for you guys probably,
I'd like to serve 10 parallell incoming calls at the same time, so I bought a
lot of Zap-channel cards for analog phone lines.
But I want all users to be able to use the same phone number to dial in, but I
want the number to be switched to an avaiable
Hi,
I am using RT engine to log queue_log to a mysql database. My extconfig is
[settings]
queue_log = mysql,asterisk16_production
Logging to mysql is working fine.
But I find that the queue_log file now only has QUEUESTART lines for eg:
1250519094|NONE|NONE|NONE|QUEUESTART|
Just tell your telco to put your numbers in a hunt group so a,b,c,d,e, etc
all come in as a.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Johan Sandgren
Sent: Monday, August 17, 2009 9:44 AM
To: Asterisk Users Mailing
You rite Kevin,
We enabeled sec echo canceller. I'll be test now and let's know the results.
Regards,
On Tue, Aug 18, 2009 at 8:47 AM, Kevin P. Flemingkpflem...@digium.com wrote:
Luis Morales wrote:
The error on system crash is:
Digum Board: TDM2400P
OS: Debian Lenny 5.02
That's a problem that needs to be discussed with the provider of those POTS
lines..
With POTS lines, each number has a unique number, but the telco builds a hunt
group for you..
For example, maybe you own phone number 555-1000. Your POTS lines may have
555-1000, 555-1001, etc, all the way to
Thanks! It's exactly what I'm looking for!
Great :)
/Johan
-Ursprungligt meddelande-
Från: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] För Steve Jones
Skickat: den 17 augusti 2009 16:52
Till: asterisk-users@lists.digium.com
Ämne:
Luis Morales wrote:
You rite Kevin,
We enabeled sec echo canceller. I'll be test now and let's know the results.
SEC is not a good choice. If you are going to try something other than
HPEC, use MG2 or KB1, which are the current best options that are
included with DAHDI. You can also stick
Hello John ,
On Mon, 17 Aug 2009, Lee, John (Sydney) wrote:
Thanks Tilghman.
I learnt it the hard way - I never imagined I need to jot down the
serial number of a PCI card :-(
If you still have the paper work from the box that came to you . The
stock agent , if you are lucky
Keving,
We use MG2 and KB1, but the best result was on SEC. How i can do to
modify dahsi to include SEC option with generic CPU.
Jhon Lee,
Take a look on this link, there are an option to solve your issue:
http://archives.free.net.ph/message/20080126.111546.a2569851.nl.html
Regards,
On Tue,
Luis Morales wrote:
Keving,
We use MG2 and KB1, but the best result was on SEC. How i can do to
modify dahsi to include SEC option with generic CPU.
If you are not using HPEC, then none of this matters; when you use an
echo canceller included with DAHDI, it's compiled for your CPU type and
Hi guys..
I just wanted to know if this config could work correctly, since a lot of u
guys have been working a lot with asterisk...
Asterisk 1.4.21.2 - chosen because is the last with Zaptel support ?
Zaptel 1.4.12
Add ons 1.4.9
Currently i have a 1.6.1.0 Server running with no problems, but
On Mon, Aug 17, 2009 at 11:18:55AM -0500, Ismael Ruiz wrote:
Hi guys..
I just wanted to know if this config could work correctly, since a lot of u
guys have been working a lot with asterisk...
Asterisk 1.4.21.2 - chosen because is the last with Zaptel support ?
Zaptel 1.4.12
Add ons
Hi all,
When I have 2 masks that would like to execute the same logic, there is
the way to use the Goto (or any other) command without changing the
${EXTEN}?
Eg. DID range is 1200-1349 - call Macro(disca), what mask to use? (I just
got it with 2 masks, but I didn't wanted to duplicate the
will exten = _13[0-4]X,1,Goto(12${EXTEN:3},1) work for you?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Aug 17, 2009, at 9:58 AM, Gabriel Ortiz Lour wrote:
Hi all,
When I have 2 masks that would like to execute the same logic,
there is the way to use the
No Jim,
It will fall on the same case, where when I dial 1349 - 1249 will get
called
I think that I will have to use another variable than ${EXTEN} for this...
2009/8/17 Jim Dickenson dicken...@cfmc.com
will exten = _13[0-4]X,1,Goto(12${EXTEN:3},1) work for you? --
Jim Dickenson
How does one go about accessing gosub arguments from Asterisk in
extensions.lua?
For example, I have the following in extensions.conf:
exten = 1000,1,Wait(1)
exten = 1000,n,Gosub(functions,mytest,1(123))
exten = 1000,n,Hangup
And then the following in extensions.lua:
extensions = {
Testing I have seen that there is data on the channel structure that tells
me the dialed number even after the Goto:
-- Executing [1...@ramais-internos:1] Goto(SIP/pa01-083308d0,
1249|1) in new stack
-- Goto (ramais-internos,1249,1)
-- Executing [1...@ramais-internos:1]
Srry!
I'd should have searched befor asking, just found it on voip-info:
*From asterisk-1.4.20.1.tar.gz; docs/channelvariables.txt:*
${DNID} - Dialed Number Identifier (Deprecated; use
*${CALLERIDhttp://www.voip-info.org/wiki/view/Asterisk+func+callerid
(dnid)}*)
*${CALLERID
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
I'm trying to connect to ekiga.net through a client connected to my
Asterisk server. For it I am being based on this [1] document. Next I
put the configurations that I am using.
/etc/asterisk/sip.conf:
; Outgoing to ekiga.net
[ekiga]
Hi,
Here is my problem. I am trying to get the Status of the call if the user
picked up the phone or not. It is coming as empty. Please help.
Here is my extensions_additional.conf file code:
[multi-dir-callback]
include = multi-dir-callback-custom
exten = _X.,1,Answer
exten =
I recently upgraded my Asterisk system to Dahdi and now I have an echo
problem.
I am running Asterisk 1.4.25 with Dahdi Complete 2.2.0, on a Digium
TE121B PCI express card with a HARDWARE echo cancellation module. All
this is housed on a CentOS 5.5 box, 2.6.18 Kernel. My incoming phone
Daniel,
Check your stunaddr setting. Is it misspelled, or do they really use
stun.exiga.net instead of stun.ekiga.net ?
N.
Daniel Bareiro wrote:
Hi all!
I'm trying to connect to ekiga.net through a client connected to my
Asterisk server. For it I am being based on this [1] document. Next I
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
SIP wrote:
Daniel,
Hi SIP.
Check your stunaddr setting. Is it misspelled, or do they really use
stun.exiga.net instead of stun.ekiga.net ?
Thanks to indicate that error to me. I doing the test again. I don't believe
that this solves what I
Not sure if anyone has mentioned this on the asterisk list yet or not.
Came across this project called SipDroid today - http://sipdroid.org/
Sipdroid is an open-source SIP client implemented in Java. The project
was based on:
* Mjsip http://mjsip.org contributing the original
On Monday 17 August 2009 12:24:08 pm Brian Camp wrote:
How does one go about accessing gosub arguments from Asterisk in
extensions.lua?
You cannot. The various methods of dialplan creation are not designed to be
interoperable. Some people have made various methods work (such as between
Not sure if anybody noticed, but it seems like Skype For Asterisk is out.
$66 per channels, pretty pricey
http://store.digium.com/productview.php?product_code=1SFA0001
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon
I am using the beta and its pretty good for remote access for clients
It would help if they had some discount structure for volume
Cheers Duncan
Pascal Bruno wrote:
Not sure if anybody noticed, but it seems like Skype For Asterisk is out.
$66 per channels, pretty pricey
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