Hi all
Do any of you know i can force asterisk to lookup ip
addresses for peers and
trunks everytime it tries to make a call.
One of the peers has a dynamic ip and is using DynDNS
to register host. Now
i need to reload asterisk everytime i want to call it
thanks
liaan
___
On 27/01/2005 03:38 Eric Wieling said the following:
Vahan Yerkanian wrote:
Was wondering if there are any news on the native MoH patch for
1.0.3/1.0.5.. or this still works on CVS HEAD only?
Since 1.0.x is for bug fixes only and not new features, I doubt that
patch will ever be in 1.0.x.
i'm u
On Wed, 2005-02-02 at 21:02 -0800, Robert Howard wrote:
> I am trying to use the MYSQL command to insert a
> record into a database and I can't seem to get it to
> work. I can do an UPDATE with no problem.
> Here is the line in my dialplan
> exten => s,12,MYSQL(QUERY resultid ${connid} INSERT
> INT
Hi, asterisk gurus:
My purpose is to key in some number after the call is
finished. The number keyed in will be stored in the
database with the phone number dialed. But whenever a
key is pressed in/after h extesion, asterisk exits the
call flow.
Does anybody has a solution for this? Is DEADAGI is
On Feb 3, 2005, at 4:20, Matthew Boehm wrote:
I'm trying to stay away from a software based load balancer cause what
happens if that server fails?
Its far less likely for a piece of dedicated hardware to fail than an
actual
computer.
There are useful things like "heartbeat" which can transparently
> The PSTN lagging would make sense and would my CDR reccord still show
> that 5911079 was dialed?
Yes, the CDR won't show the w
HTH
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
T
Altus Snyman wrote:
Good day all
I downloaded bristuff RFC3 and asterisk,zaptel,libpri versions 1.0.3
This is to install my quad bri card
All installed well
I coped over some old config files.All 4 ports are available,so that
gives 8 open lines for incoming or outgoing,correct me of I'm rond
The pr
[EMAIL PROTECTED] wrote:
Hello,
I have looking into the TDM series of wildcards.
All these card are for linux kernel 2.4.
If I were to use FC3 which is based on kernel 2.6, will
I have any compatibility issues.
Thanks
I'm not sure about Fedora, but we're running SuSE 9.1 with the 2.6 kerne
Good day all
I downloaded bristuff RFC3 and asterisk,zaptel,libpri versions 1.0.3
This is to install my quad bri card
All installed well
I coped over some old config files.All 4 ports are available,so that
gives 8 open lines for incoming or outgoing,correct me of I'm rond
The problem is,asterisk ca
The PSTN lagging would make sense and would my CDR reccord still show
that 5911079 was dialed?
On Wed, 2 Feb 2005 15:29:23 -0800 (PST), Miguel Ruiz Velasco Sobrino
<[EMAIL PROTECTED]> wrote:
> It may be a problem of the PSTN not catching the initial 5
>
> change your dial string from
> DIAL(zap/
Hello,
I have looking into the TDM series of wildcards.
All these card are for linux kernel 2.4.
If I were to use FC3 which is based on kernel 2.6, will
I have any compatibility issues.
Thanks
Varun
___
Asterisk-Users mailing list
Asterisk-Use
Would you be kind enough to share your Sipura 3000 setup? I've got
mine passing calls
to and from Asterisk, I just can't get it pass AU caller ID (I always
get the sipura's extension as the caller ID)..
On Mon, 31 Jan 2005 22:18:31 +1000, Peter Illmayer <[EMAIL PROTECTED]> wrote:
> Nathan
>
tuning for ulaw g.711 - Polycom IP500
I've got QoS ironed out at this point
(dedicated section of bandwidth with VoIP having priority), yet ulaw is
still unusable due to small 'holes' in the audio. Running iperf tests shows
packet loss at very low percentages (< 0.02 %), and when we do drop,
it's
I am trying to use the MYSQL command to insert a
record into a database and I can't seem to get it to
work. I can do an UPDATE with no problem.
Here is the line in my dialplan
exten => s,12,MYSQL(QUERY resultid ${connid} INSERT
INTO `member` ( `id` , `member_num` , `active` )VALUES
('',${number}' ,
I had my [EMAIL PROTECTED] server working fine SPA-8841
SPA-2100. It was on an open IP no fire wall. I moved the server
behind the firewall. Now the phones will not dial out. The phones
can be called from a DID or calling to the main POTS number and dialing the extension.
However neith
> I'm trying to stay away from a software based load balancer cause what
> happens if that server fails?
> Its far less likely for a piece of dedicated hardware to fail than an actual
> computer.
You really ought to open up one of those pieces of dedicated hardware
sometime and see what's inside.
> LVS is a single point of failure, but probably so is your router/switch.
> Consider the case where the LVS *is* the router, use good quality
> components for the PC (we should all know about this part on this list),
We've used the via-based eden motherboards for this sort of thing - rock
solid,
Est ce que quelqu'un peut me dire pourquoi le volume de mes enregistrement
sont tout le temps trop bas.
Hi,
I am trying to figure out why my recorded files have a very low volume ? I
tried gsm, wav with no success.
___
Asterisk-Users mailing list
Asteri
On Wed, 2005-02-02 at 21:20 -0600, Matthew Boehm wrote:
> I'm trying to stay away from a software based load balancer cause what
> happens if that server fails?
> Its far less likely for a piece of dedicated hardware to fail than an actual
> computer.
There are many ways to accomplish this, and th
Pedro
Exactly my point. I have each company in a different
context. How do I SetCallerID to a number based on the context they are in?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listin
On Wed, 2005-02-02 at 15:55 +0100, Edgar de Leon wrote:
> hello i need to know how to enable the feature in the agents.conf to make
> the users got to press # to answer the call when is in the queue and the
> agent is logged in.
>
> at this time the call enters the queue and the agents who is logg
When you purchase your G.729 licenses from
http://www.digium.com/index.php?menu=asterisk_g729 instructions will
be included, including the download location. If you have trouble
installing your licenses you should contact digium technical support.
On Wed, 2 Feb 2005 17:56:22 -0800, Ing. Ignacio O
I'm trying to stay away from a software based load balancer cause what
happens if that server fails?
Its far less likely for a piece of dedicated hardware to fail than an actual
computer.
-Matthew
- Original Message -
From: "Kyle Loree" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing Lis
I'm trying to get into the world of Asterisk
in order to use the voicemail and autoattendat features (and more stuff later)
with a Redcom switch. But, I've only started and haven't gotten to that yet. At
this point my solitary goal is to talk to the autoattendant via an SIP phone
(SJPhone).
You need to create different contexts for each company.
- Pedro
On Wed, 2 Feb 2005 21:49:53 -0500, Jason Brown <[EMAIL PROTECTED]> wrote:
>
>
>
> In order to put a shared pbx in an office building for multiple businesses,
> I will have to make sure that the caller ID information going out i
In order to put a shared pbx in an office building for
multiple businesses, I will have to make sure that the caller ID information
going out is correct.
i.e. company a’s main phone number is 5551212
company b is 5572121
company c is 5596767
Now I know how to distribute inco
Hello everyone
can anyonone of you send me the file codec_g729.so this file has to be
inserted in
/urs/lib/asterisk/modules
Thank You
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-user
Mike Sander wrote:
Hi
I know that attended transfers are only available in the CVS Head.
The version info reports:
Asterisk CVS-v1-0-02/03/05-10:24:22
You don't have HEAD. Follow the instructions on asterisk.org to
download from CVS HEAD (as in, not -r v1-0).
Nick
_
Hi
I know that attended transfers are only available in the CVS Head.
I downloaded the asterisk-update.sh script from voip-info.com and ran it
with these parameters
./asterisk-update.sh update dev
It looked as tho CVS HEAD was downloading and compiling, although it
couldn't download the addons.
DNS based load ballancing has it's place, as dose using an
application level switch.
Say an earthquake takes out your California data center.
Shortly the DNS servers will notice and pull that center's
record. However do to caches and all this is not fast
and users will notice.
What the switc
Sergey,
I am in Windsor ... That is 3.5 hours away ...
Btw: how much is colo there?
Robert
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sergey
Kuznetsov
Sent: Wednesday, February 02, 2005 7:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discus
Robert,
Honestly, it's better to get colo at 151 Front St. from any big ISP
company.
Robert Augustyn wrote:
Hi,
Have you tried it?
Any comments would be greatly appreciated.
I can have it at C$200, is that a good price?
Thanks a lot.
robert
___
Asterisk-U
For the brave, you can test it out yourself (if you can get the beta to
work without documentation) with the callgenerator on
http://www.astertest.com/forum/viewtopic.php?t=4
Its far from finished, but it can be used.
zoa.
Steven Critchfield wrote:
On Wed, 2005-02-02 at 18:51 +0100, Roy Sigurd Karl
I've got mostly Cisco 7960s and a few Analog phones
on TDM Ports. On the 7960s, the echo is quite bad.
On the TDM ports, it is there, but not as bad. I have
tried setting echo cancellation to various numbers, but
have had no luck.
This began after a HEAD version of * was installed.
Since th
How about a management server that polls the asterisk servers every
minute with snmp to check cpu and ram cache, maybe even drive space.
Then you could have a script decide whether the server can handle
anymore connections.
I am still a beginner so I am not sure how you could have asterisk
de
I have a problem with my [EMAIL PROTECTED] installation
(configured with AMP)
My question is this, can you have different ports for
different contexts within IAX?
[Faktortel]
port = 5036 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to
bind to
context = default ; Default for
inc
On Feb 3, 2005, at 0:03, Miguel Ruiz Velasco Sobrino wrote:
--- [EMAIL PROTECTED] wrote:
The DNS approach does not handle single or multiple system failures,
only very elementary load balancing over a lengthy period of time.
Are you shure of that? I'm aware that the load criteria is trickier,
but
Hi
I know that attended transfers are only available in the CVS Head.
I downloaded the asterisk-update.sh script from voip-info.com and ran it
with these parameters
./asterisk-update.sh update dev
It looked as tho CVS HEAD was downloading and compiling, although it
couldn't download the addons.
It may be a problem of the PSTN not catching the initial 5
change your dial string from
DIAL(zap/1/${EXTEN})
to
DIAL(zap/1/ww${EXTEN})
note a "w" before the number, each w makes a 1/2 sec pause and you can put many
of them,
so if your PSTN lags a little to give you dialtone, that probably make
Andrew,
What happens when you dial other numbers?
Is it stripped on those as well?
Can you look in your zapata.conf for stripmsd=1 ?
Kyle
On Feb 2, 2005, at 3:51 PM, Andrew Niemantsverdriet wrote:
I figured out how to view it. Here is what it says:
# cat /var/log/asterisk/cdr-csv/Master.csv | grep
On Wed, 2 Feb 2005 15:51:53 -0700, Andrew Niemantsverdriet
<[EMAIL PROTECTED]> wrote:
> So it looks to me like something else went wrong.
If you took your dial line right from the samples you likely still
have a ${EXTEN:{TRUNKMSD}}
That variable TRUNKMSD is probably stripping off the first digi
On 15:29, Wed 02 Feb 05, Leif Madsen wrote:
> kernels. I believe it also needs to be 2.4.9 or later (but who's still
> using kernels that old, honestly :))
>
What's wrong with 2.0.40
Works like a charm on some of my routers and firewalls.
--
Michiel van Baak
http://lunteren.vanbaak.info
[EM
--- [EMAIL PROTECTED] wrote:
> The DNS approach does not handle single or multiple system failures,
> only very elementary load balancing over a lengthy period of time.
Are you shure of that? I'm aware that the load criteria is trickier, but very
possible.
If you use DDNS (dynamic DNS) using B
> According to Volume 1 Asterisk Docs
>
> To create an FXO channel on the same TDM400P card, we list all the
> settings for the channel and then define the channel number. Instead of
> only having signalling be fxo_ks though we want the signalling to be
> fxs_ks. Because the other settings haven't
CPU. 729 transcoding is CPU heavy.
-Matthew
- Original Message -
From: "Kanuri, Seshu (Company IT)" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, February 02, 2005 4:23 PM
Subject: RE: [Asterisk-Users] Linksys PAP2 / RT31P2 + multipl
I figured out how to view it. Here is what it says:
# cat /var/log/asterisk/cdr-csv/Master.csv | grep "911"
"","2000","5911079","from-sip-internal","""Andrew Niemants""
<2000>","SIP/2000-a509","Zap/1-1","Hangup","","2005-02-02
14:24:05","2005-02-02 14:24:08","2005-02-02
14:24:57",52,49,"ANSWERED",
On Wed, 2005-02-02 at 23:33 +0100, Bruno Hertz wrote:
> On Wed, 2005-02-02 at 14:21 -0800, Aaron Glenn wrote:
> > That's the least ambiguous subject I could muster. I'm relatively new
> > to Asterisk and while I'm certain there is a way to do this, I'm
> > unsure how. My question is this: How do I
On 2 Feb 2005, at 22:14, Matthew Boehm wrote:
eh..
they told me it was a CPU problem. that v1 of the pap2na didn't have
the
horsepower to transcode 2 channels.
anyone know if the Sipura 2100 can do 2 simul 729 calls?
Yes, the SPA-2100 can do two simultaneous G.729 calls. Support for this
was
According to Volume 1 Asterisk Docs
To create an FXO channel on the same TDM400P card, we list all the
settings for the channel and then define the channel number. Instead of
only having signalling be fxo_ks though we want the signalling to be
fxs_ks. Because the other settings haven't been change
On Wed, 2005-02-02 at 14:21 -0800, Aaron Glenn wrote:
> That's the least ambiguous subject I could muster. I'm relatively new
> to Asterisk and while I'm certain there is a way to do this, I'm
> unsure how. My question is this: How do I take an incoming call, put
> the person on hold, and in the ba
Being a Newb I don't know how to look at my CDR, could you tell me.
On Wed, 2 Feb 2005 16:21:10 -0500, Andrew Kohlsmith
<[EMAIL PROTECTED]> wrote:
> On February 2, 2005 04:15 pm, Andrew Niemantsverdriet wrote:
> > I can see why I think; 5 "911" 079. But I don't understand why it is
> > being hand
That looks like what I have...
Jim Richards
Computer Security Officer
Wisconsin Dept of Transportation
-Original Message-
From: Martin Roy [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 02, 2005 3:39 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] RE: Incoming calls
On Sun, 30 Jan 2005 00:29:53 +0100, Philipp von Klitzing
<[EMAIL PROTECTED]> wrote:
> Hi!
>
> > The transfer is always successful, but sometimes (lets say bout 50% of
> > the times), the MusicOnHold does not "de-attach" itself from the call,
>
> Maybe a good start for an investigation is to try a
-Original Message-
eh..
they told me it was a CPU problem. that v1 of the pap2na didn't have the
horsepower to transcode 2 channels.
anyone know if the Sipura 2100 can do 2 simul 729 calls?
-Matthew
/Snip/
Horse Power?? Please elaborate. Is it a CPU issue or a power issue or
onboard Ram i
That's the least ambiguous subject I could muster. I'm relatively new
to Asterisk and while I'm certain there is a way to do this, I'm
unsure how. My question is this: How do I take an incoming call, put
the person on hold, and in the background (i.e. while they are on
hold) begin trying other pho
eh..
they told me it was a CPU problem. that v1 of the pap2na didn't have the
horsepower to transcode 2 channels.
anyone know if the Sipura 2100 can do 2 simul 729 calls?
-Matthew
- Original Message -
From: "Matt Klein" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commerc
I'm running into a bit of a problem setting up conference calls. The box
I rent at a colo doesn't seem to have USB hardware When I try to load
usb-uhci I receive a "device does not exist" error. Which means I can't
load ztdummy
The system has a rtc clock module, so zaprtc won't work... (
yep, post your conf.
On Wed, 2 Feb 2005, AJ Grinnell wrote:
post your dialplan from extensions.conf
On Wed, 2 Feb 2005 14:15:28 -0700, Andrew Niemantsverdriet
<[EMAIL PROTECTED]> wrote:
Hi,
I am quite new to asterisk so I am not sure what is needed to figure
out this problem. If more information is
edit /etc/asterisk/logger.conf
uncomment one of the console lines
and reload
chow
L
- Original Message -
From: "Robert Augustyn" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
Sent: Wednesday, February 02, 2005 10:36 PM
Subject: [Asterisk-Users] [EMAI
Given that you know you left pieces out, it would appear you are right
on track.
> So if I understand well this should do the trick : (be aware that
> context first and second include all my extensions that I haven't
> included in this and in my SIP phones use context fi
post your dialplan from extensions.conf
On Wed, 2 Feb 2005 14:15:28 -0700, Andrew Niemantsverdriet
<[EMAIL PROTECTED]> wrote:
> Hi,
> I am quite new to asterisk so I am not sure what is needed to figure
> out this problem. If more information is needed and not provided I
> will gladly provide it.
Hello All, I sign up with $5.99 broadvoice plan. I made in and outbound
calls OK. I upgraded to unlimited world and now I have problems with
outbound calls. I called broadvoice and they said they would get back it
me.
Here are my sip and extension files.
sip.conf
register => XX:[EMAIL P
of course it won't. neither can the ata.
they're cheap, it was a licensing decision.
i look forward to v2.
-m
On Wed, 2 Feb 2005, Matthew Boehm wrote:
Holy Crap
I have just verified this! The linksys PAP2-NA will NOT SUPPORT 2
SIMULTANEOUS G729 CALLS!
And I just got off the phone with some supe
So if I understand well this should do the trick : (be aware that
context first and second include all my extensions that I haven't
included in this and in my SIP phones use context firstinternal and
secondinternal)
zapata.conf :
context=firstincoming
switchtype=national
signalling=fxs_ks
echot
looks like an ignorepat problem on the first *number* (single) dialed
(i.e., trying to ignore the number 9 on an outbound call.)
try to make a call to 591-2079.
-
"Yeah, we rocked the vote all right. Those little
bastards betrayed us again."
- H
Title: Speex pass through on SIP
Hi,
I've seen some answers to this on the mailing list archives but nothing that seems like the right answer. What I want is for 2 SIP phones to use speex to talk to each other through 2 asterisk boxes (linked over IAX2) while only supporting ulaw on the ast
Try dialing 591-2079 and see if you're trying to make a call to "91-2079"
instead of "591-2079".
-m
On Wed, 2 Feb 2005, Andrew Niemantsverdriet wrote:
Hi,
I am quite new to asterisk so I am not sure what is needed to figure
out this problem. If more information is needed and not provided I
will g
On February 2, 2005 04:15 pm, Andrew Niemantsverdriet wrote:
> I can see why I think; 5 "911" 079. But I don't understand why it is
> being handled this way. Can somebody offer me some guidance on how to
> get this to stop?
Your FXO card missed the '5', that's all. Or maybe Asterisk did. Or mayb
submit a bug in bug tracker at http://bugs.digium.com
On Wed, 2 Feb 2005 21:55:16 +0100, Robert Rozman <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I've spotted weird crash of Asterisk cvs Stable. I have defined queue in
> queues.conf :
>
> [prodaja]
> music = default
> announce = queue-markq
> strateg
Can you post your "outbound" portion of your extensions.conf?
---
Kelly D Griffin
Network Engineer
Tantella Wireless
http://tantella.com
800.636.0306 Voice
479.464.8998 Fax
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Niemantsverdriet
Sent: Wed
Make sure your 911 match statement in extensions.conf starts with an "_"
underscore.
tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Niemantsverdriet
Sent: Wednesday, February 02, 2005 4:15 PM
To: Asterisk-Users@lists.digium.com
Subject: [Aste
Fix your dial-plan in extensions.conf so that the first digit isn't
getting dropped.
If you're using a traditional dial plan, you press 9 to get out so
many default configs will drop the first digit when sending the string
out to the carrier. You appear to be getting outside without the 9 so
the
Andrew,
can you post your extensions.conf to the list please?
Kyle
On Feb 2, 2005, at 2:15 PM, Andrew Niemantsverdriet wrote:
Hi,
I am quite new to asterisk so I am not sure what is needed to figure
out this problem. If more information is needed and not provided I
will gladly provide it.
I have a
Matthew, i think it would
be convenient that you use dns round-robin for load balancing, registering the
clients
against Ser or Asterisk boxes. Greetings.
Ariel.
- Original Message -
From: "Matthew Boehm" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial
Discus
Hi,
I am quite new to asterisk so I am not sure what is needed to figure
out this problem. If more information is needed and not provided I
will gladly provide it.
I have a very basic asterisk setup. 1 x100p card and a grandstream
handytone 286. I can make calls fine to most phone numbers from th
Hello,
Mark has changed app_dial.c in CVS-HEAD to allow for a 'p' dial flag that
will allow for the original callerid to not be altered. If you are using
CVS-HEAD, simply put the p flag at the end of your Dial string in your
extensions.conf file for 91NXXNXX Dial TRUNK step(or wherever you dia
Holy Crap
I have just verified this! The linksys PAP2-NA will NOT SUPPORT 2
SIMULTANEOUS G729 CALLS!
And I just got off the phone with some super-level technician at linksys and
he said they knew this all along!!
What bastards!
Anyway, he told me they are comming out with the PAP2-NAv2 in a
Hi,
I've spotted weird crash of Asterisk cvs Stable. I have defined queue in
queues.conf :
[prodaja]
music = default
announce = queue-markq
strategy = ringall
context = from-pstn
timeout = 15
retry = 5
maxlen = 0
announce-holdtime = no
announce-frequency = 30
announce-holdtime = yes
monitor-forma
You don't need to enable it. It happens automatically.
Log into console
With username : root
Password :password
then enter "asterisk -r"
To get out type "exit."
What happens when you enter "asterisk -r?"
If you type "sip show peers" what happens?
-Original Message-
From: [EMAIL PROT
Dan Adams wrote:
> Are you going to be making this one available to all. I am not sure if
Yes.
> or how it is possible, but maybe you would be able to have it so that
if you right click on the contact, it has an option to iniate a call from
Well, you drag the contact from outlook onto one of the
Hi Everybody,
I'm trying to make my asterisk dial a international call from a SER
request of it. My ambient is like this.
[Clients]--[SER]--[Asterisk]--[Go2Call]
Client: My SIP clients.
SER: My REGISTRAR/Proxy Server
Asterisk: All other services(Voicemail,musiconhold etc) and also act
> I beleive what you're looking for is a scalable SIP proxy, like SER :)
> That way, all clients registers to SER and SER redirects the caller to
> one of the asterisk boxes. Search the wiki at voip-info.org for
> "asterisk at large" :)
Yes, that is one of the many pages I've read. But we stil
izo wrote:
On Sat, 29 Jan 2005 11:13:36 -0500, Jim Van Meggelen wrote:
does anyone out there made some experience with Varion
(www.govarion.com) based E1/T1 cards ?
Their cards work. The only problem about govarion is their delivery
time. The cards are just not shipped as promised. And it
Hi,
I am connecting to the asterisk using asterisk -r command but I never get
anything on the console?
How can I enable it?
Robert
Btw: it is version 0.4
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/l
On Wed, 2 Feb 2005 15:34:24 -0400, Daniel del Castillo
<[EMAIL PROTECTED]> wrote:
> I'm having problems trying to run zaptel. I don't have the hardware, I
> first want to test out asterisk. The problem is the usb-uhci/usb-ohci
> module, it isn't present on the system as same as usbcore and I don't
> OK I have 12 phone lines connected to 3 digium TDM04B cards on the same
> server. I must do the following thing :
>
> The first 10 lines will be use by one company and the 2 left by another
> one. For outgoing calls it's quite easy I just create 2 different group
> and let them dial on a dif
On Wed, 2005-02-02 at 18:51 +0100, Roy Sigurd Karlsbakk wrote:
> >> 6b- More than 50 calls VoIP to POTS/T1/E1 will kill an * box due to
> >> excesive transcoding??
> >> 6bb- unless using quad machines, plenty of RAM and DSP cards?
> >
> > The max number of calls is something that's not real
On Thu, 2005-02-03 at 07:07, Martin Roy wrote:
> OK I have 12 phone lines connected to 3 digium TDM04B cards on the same
> server. I must do the following thing :
>
> The first 10 lines will be use by one company and the 2 left by another
> one. For outgoing calls it's quite easy I just create 2
Martin,
You would want to put them into different contexts (in the zapata.conf).
Using different contexts you can slice and dice up your channels to your
hearts content.
You would then be able to have an S,1,Answewr in that context.
Jim
-Original Message-
From: Martin Roy [mai
Martin Roy wrote:
OK I have 12 phone lines connected to 3 digium TDM04B cards on the
same server. I must do the following thing :
The first 10 lines will be use by one company and the 2 left by
another one. For outgoing calls it's quite easy I just create 2
different group and let them dial on
Hmm i found the problem... I´m using a Grandstream BT100. The transfer just
works in a queue if I first acknowledged the call using the # key, and then
press the TRANSFER key in the Grandstream.
In the asterisk console I receive a:
-- SIP/4002-4563 acknowledged
Then I can transfer the call...
OK I have 12 phone lines connected to 3 digium TDM04B cards on the same
server. I must do the following thing :
The first 10 lines will be use by one company and the 2 left by another
one. For outgoing calls it's quite easy I just create 2 different group
and let them dial on a different one. B
Sorry forget my post I forgot to add it in zaptel.conf... now it's
working fine... That's what happen when you want to do things a little
too fast hehe ;-)
Martin
Martin Roy wrote:
I already have one Digium TDM04B installed in my server working fine.
I just received 2 more so I did added them t
I already have one Digium TDM04B installed in my server working fine. I
just received 2 more so I did added them to my server. When I booted
again linux told me that it found new hardware I said ignore. Then I log
in as usual. I did ztcfg -vv to see if it sees 12 channels now instead
of 4 but i
Em Qua 02 Fev 2005 17:34, Daniel del Castillo escreveu:
> I'm having problems trying to run zaptel. I don't have the hardware, I
> first want to test out asterisk. The problem is the usb-uhci/usb-ohci
> module, it isn't present on the system as same as usbcore and I don't
> know why. Any tip?
Do y
I'm having problems trying to run zaptel. I don't have the hardware, I
first want to test out asterisk. The problem is the usb-uhci/usb-ohci
module, it isn't present on the system as same as usbcore and I don't
know why. Any tip?
--
-DdC
___
Asterisk-Us
Ronald,
First step, try switching the modules around on your card.
I had the same issue and had to rma the module.
Kyle
On Feb 2, 2005, at 12:07 PM, Ronald Hartmann wrote:
Erros on boot. Running Release 1 Latest Version.
Feb 2 13:32:20 localhost kernel: Zapata Telephony Interface Registered
on maj
Erros on boot. Running Release 1 Latest Version.
Feb 2 13:32:20 localhost kernel: Zapata Telephony Interface Registered
on major 196
Feb 2 13:32:20 localhost kernel: Freshmaker version: 71
Feb 2 13:32:20 localhost kernel: Freshmaker passed register test
Feb 2 13:32:29 localhost kernel: Module
Is there a way to have a caller go back to the operator once they are in the
voicemail directory or are they stuck?
IE I call in and don't' know the extension but go to the company directory
and can't find who I want, how do I get back to the operator?
Sometime ago, I wrote an example of a functional queue scenario.
Perhaps you give it a try.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Queue
Btw, how is the queue command invoked in your extensions.conf?
Post your relevant sections of queues.conf, agents.conf and extensions.conf
1 - 100 of 212 matches
Mail list logo
