On Tue, 2005-10-25 at 21:27 +0200, Louis-David Mitterrand wrote:
Hi,
I'm trying to get a SPA-3000 to work with a Siemens Gigaset 3010 DECT
(cordless) phone. I tried every localization scheme I could find on the
Net, including the settings recommended by the Voxilla wizard.
This Gigaset
On Tue, 2005-10-25 at 21:27 +0200, Louis-David Mitterrand wrote:
Hi,
I'm trying to get a SPA-3000 to work with a Siemens Gigaset 3010 DECT
(cordless) phone. I tried every localization scheme I could find on the
Net, including the settings recommended by the Voxilla wizard.
This Gigaset
marek cervenka wrote:
hi,
will be somewhere materials (videos, presentations) from astricon?
Registered attendees will get information about the material soon.
No videos where recorded this year.
The 1.2 presentation I made together with Kevin has been available
for a while at
Forget about writing perl scripts, just use SipSak (SIP Swiss Army Knife) to
send a SIP registration (including authentication if you want) and check
the return value ($?)
Benjamin Lawetz wrote:
Damn, so many things left to learn :-)
Thanks
-Original Message-
From: [EMAIL
I've a strange problem here. I can dial out via an AVM B1 card.
I have a sip client running. I can hear my conversational partner
but he can't here me. I'm using * 1.0.
For SIP and CAPI operation there is no soundcard required at the
asterisk server.
Perhaps your SIP client does require
Sipsak could be used to test the SIP registration.
Asterisk redundancy is a very interesting topic.
I also thought about trying to run two asterisk boxes with VRRP
(Virtual router redundancy protocol), and usign the virtual IP address created
by VRRP as the SIP registration address.
Andrew Kohlsmith napisał(a):
Oct 25 16:53:56 WARNING[29761]: chan_sip.c:4826 check_auth: Stale nonce
received from 'sip:[EMAIL PROTECTED];user=phone'
First off, it's a warning. It's not a bad thing. It should be relegated to
higher debug status, IMO since it's just informational.
In my
Hi,
I have an AAH installation with an active Eicon DIVA BRI card. My AAH is
built on Centos 3.5 which is at kernel
2.4.21.37.EL.
I have installed the source level RPM from Eicon as well as chan_capi-0.3.5.
When I try to run divactrl load -c 1 -f ETSI -Debug I get a response:
A: can't get card
I've TDM400P with 2 cards fxo and asterisk 1.0.9 + zaptel 1.0.9.2
All works perfectly but command Hangup or Hangup() in dialplan dont
hangup call
(zapata.conf within busycount=4 and busydetect=yes)
Why ?
Country is ITALY
--
Giovanni Miano
___
Hello,
I have trouble getting asterisk to work with my new
firewall script (see below).
I used this info as base: 'http://www.voip-info.org/wiki-Asterisk+firewall+rules
And then modified it to suit my needs.
I use only SIP and the problem is that the calls
get in to asterisk when the
Hi,
I saw (and it's works), that you can mix the realtime and static mode.
In extconfig.conf file configure to use sip.conf in realtime
...
;realtime
sipusers = mysql,pbx,PBX_sip_buddies
sippeers = mysql,pbx,PBX_sip_buddies
...
Don't delete the sip.conf file!
In the sip.conf file define only
Hi,
I have been doing some tests with app_meetme, all the clients i used
were SIP clients, and i have noticed that MeetMe continues to decode
the channels of the clients even if they are just connected to a
conference as listeners or muted. This really affects the performance
of Asterisk since
Hi,
I'm attempting to run * as a non-root user (asterisk), I can run * as
my new user with /usr/sbin/asterisk -c without problem.
However, I'm unable to run * using safe_asterisk with my user, the error
shown is:
Asterisk ended with exit status 127
safe_asterisk is trying to
Hi!
I'm running asterisk as a non-root user without problems. I followed the
instructions on
http://www.voip-info.org/tiki-index.php?page=Asterisk%20non-root
If it still does not work, append the last part of your
/var/log/asterisk/messages
Christian
gincantalupo schrieb:
Hi,
I'm
On Wed, 26 Oct 2005 [EMAIL PROTECTED] wrote:
Hi,
I have an AAH installation with an active Eicon DIVA BRI card. My AAH is
built on Centos 3.5 which is at kernel
2.4.21.37.EL.
I have installed the source level RPM from Eicon as well as chan_capi-0.3.5.
You should use new chan_capi-cm from
On Wed, 26 Oct 2005, Esteban Guana-Jarrin wrote:
Can anyone please provide some help. I have installed an AVM fritz card on an
asterisk box ([EMAIL PROTECTED] version 1.5). I have installed the card
driver and
chan_capi-cm-0.6. According to the installations guide I can now see that the
CAPI
On Wed, Oct 26, 2005 at 10:11:44AM +0530, Omkar Pandit wrote:
Hi,
I have a PPC architecture board with Linux running on top of it. I need to
get Asterisk running on it. I have the following questions:
1. What should be the version of Linux Kernel for running Asterisk? Also,
what utilities
Ive had the same problem on my boxes, and finally
had to settle for using type=peer, and defining the host=x.x.x.x despite what
the documentation says. If the sip connection cannot find either a peer or a
user, then is uses the context=? In the general section of the sip.conf
file.
Some people is still waiting for last Astricon materials; what about them ?
Regards.
Marco Vescovi
-Messaggio originale-
Da: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di Olle E.
Johansson
Inviato: mercoledì 26 ottobre 2005 8.42
A: Asterisk Users Mailing List -
Hi!
Sorry for disturbing you, I found what I was looking for.
The strange thing was I found nothing inside my
/var/log/asterisk/messages and inside /var/log/messages.
At the end, I found that safe_asterisk script wants the user to know
where asterisk is located, so I added /usr/sbin to the
Any word on the availability of the Madrid materials?
Craig
- Original Message -
From: Olle E. Johansson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, October 26, 2005 2:42 PM
Subject: Re:
Hi,
i have set up asterisk on debian sarge (kernel 2.2.6.8-2) with chan_capi
and AVM Fritz Card PCI. Asterisk starts up fine, but it does not respond
to any call on the specified MSN.
I have installed the asterisk sample configuration (make sample)
When i enable capi debugging in the asterisk
yes,
I tested too and it's works.
The Problem is when we want to add (or delete)
registers without reload the asterisk.
We are using it like a border server wich
is registered like many users in a SER machine
and the real endpoints are registered in the
asterisk.
Regards.
Jsalas
Thanks guys for the info on Sipsak, testclient, and testserver. These
will be very helpful.
___
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
On Wed, 2005-10-26 at 09:14 +0200, Joseph Rothstein wrote:
[snip]
I also thought about trying to run two asterisk boxes with VRRP
(Virtual router redundancy protocol), and usign the virtual IP address
created by VRRP as the SIP registration address. Have not had a chance
to test it, but
Hi all,
I have a shed load of UK (BT) analogue equipment at our office (18
phones, 6 faxes) hooked up via structured cabling to an old Avaya
ArgentOffice phone system. The on-its-way-out phone system has an
IDSN30e PRI interface talking to the BT exchange delivering 18 channels.
If I were
You can try this patch
(www.maxosystem.net/asterisk/asterisk-stable-polarity.html), if your
telco sends your polarity reversals on answer and hangup.
Julian J. M.
On 10/26/05, Giovanni Miano [EMAIL PROTECTED] wrote:
I've TDM400P with 2 cards fxo and asterisk 1.0.9 + zaptel 1.0.9.2
All works
On Wed, Oct 26, 2005 at 02:17:11PM +0100, Chris Shucksmith wrote:
I have a shed load of UK (BT) analogue equipment at our office (18
phones, 6 faxes) hooked up via structured cabling to an old Avaya
ArgentOffice phone system. The on-its-way-out phone system has an
IDSN30e PRI interface
Yair Hakak wrote:
Hello all,
forgive me if this is a simple question, but does bridging a SIP
channel and an IAX channel that use the same codec (say, alaw) involve
transcoding? i'm trying to figure out what kind of hardware i'll need,
and i'm going to be using SIP endpoints and IAX
Dear users,
It might be slightly off topic.
I own couple 500 and 600 Polycom SoundPoint IP phones and
need to download new software for them.
The phones has been purchased from voipsupply.com
Is there a way to download such a software without becoming certified
reseller?
Thanks,
Bartosz
I will send them to u.
Bruno De Luca
Bartosz Jozwiak wrote:
Dear users,
It might be slightly off topic.
I own couple 500 and 600 Polycom SoundPoint IP phones and
need to download new software for them.
The phones has been purchased from voipsupply.com
Is there a way to download such a
Does anyone know if SIPURA SPA-2002's support DNS SRV records?
One question I have about this for someone. If you are using PRIs or
FXO, or whatever for inbound calls. How are you doing routing?
In other words, if you dup the configs, when server A is up, how do
you route any calls that come
On 10/26/05, Matt [EMAIL PROTECTED] wrote:
Does anyone know if SIPURA SPA-2002's support DNS SRV records?
One question I have about this for someone. If you are using PRIs or
FXO, or whatever for inbound calls. How are you doing routing?
In other words, if you dup the configs, when server A
This sounds to me like a solution that DUNDi (http://www.dundi.com)
could solve. I have a rough draft of a paper on DUNDi that I started,
but which needs to ultimately be updated at
http://leifmadsen.com/papers/dundi-intro.pdf. If anyone is *really*
interested in having the paper updated and
You should also ensure the PRI is really configured for EuroISDN, many
BT PRI's are actually UK ISDN which Asterisk doesn't support (it's an
older version).
I had a problem along these lines, when I first started with asterisk, the
PRI was originally DASS2, but needed to be Q931 Full ETSI for
Following the instructions at http://geekgazette.com, I have the SPA-3000
setup as a SIP trunk. This is working flawlessly with one exception, it
isn't passing caller ID. Regardless of what settings I have tried, I can't
seem to figure this out. Has anyone else got it to work?
-Kerry
On Wed, Oct 26, 2005 at 03:33:16PM +0100, Mark Ackroyd wrote:
You should also ensure the PRI is really configured for EuroISDN, many
BT PRI's are actually UK ISDN which Asterisk doesn't support (it's an
older version).
I had a problem along these lines, when I first started with asterisk,
On Wed, 2005-10-26 at 10:12 -0400, Matt wrote:
Does anyone know if SIPURA SPA-2002's support DNS SRV records?
One question I have about this for someone. If you are using PRIs or
FXO, or whatever for inbound calls. How are you doing routing?
In other words, if you dup the configs, when
Kerry Garrison wrote:
Following the instructions at http://geekgazette.com, I have the SPA-3000
setup as a SIP trunk. This is working flawlessly with one exception, it
isn't passing caller ID. Regardless of what settings I have tried, I can't
seem to figure this out. Has anyone else got it to
On the PRI side you can use the failover equipment from e.g.
junghanns.net.
Sorry I'm not seeing failover equipment? I'm seeing PRI cards and an
ISDN guard?
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
On Wed, 2005-10-26 at 11:02 -0400, Matt wrote:
On the PRI side you can use the failover equipment from e.g.
junghanns.net.
Sorry I'm not seeing failover equipment? I'm seeing PRI cards and an
ISDN guard?
PRI
|
|
Right got it.. sorry I thought I stopped that e-mail from going out.
Very cool!
Can you give me an idea of what you do for DNS SRV to get the sip
devices to flip? Or do you just have the other asterisk server take
over the IP of the old one (seems like a good solution).
On 10/26/05, Patrick
Hi Jörg,
vielen Dank für deine Antwort. Ich denke du meinst die extensions.conf
(bin noch Anfänger, was Asterisk angeht).
Ich habe an der Datei noch nichts verändert.
So sieht sie bei mir so aus:
;
; Static extension configuration file, used by
; the pbx_config
A phone plugged into it will grab the CID on about the second ring and I
have adjusted the SPA3000 out to 5 rings with no difference. What gets
passed to asterisk is whatever is set in the 3000's Display Name field. If
the Display Name field is blank, then nothing comes across and the phones
Bartosz Jozwiak wrote:
Dear users,
It might be slightly off topic.
I own couple 500 and 600 Polycom SoundPoint IP phones and
need to download new software for them.
The phones has been purchased from voipsupply.com
Is there a way to download such a software without becoming certified
Hi there,
This'll be an easy answer for a lot of you, just wanna make sure:
We ordered a 2 phone line package from SBC in Chicago (two numbers,
one main number), pretty basic straigt forward. The line to the
building is a four wire line with an RJ-11 connector. We want to
connect both of
Paul Herman wrote:
Hi there,
This'll be an easy answer for a lot of you, just wanna make sure:
We ordered a 2 phone line package from SBC in Chicago (two numbers,
one main number), pretty basic straigt forward. The line to the
building is a four wire line with an RJ-11 connector. We
Hi ALL;
I have users with Sipura/Linksysphones
regsitered behind Nat( useing STUNat phonenot
portforwarding) in my Asterisk box, when I try to call them
with another phone i got:
Got SIP response 404 "Not Found" back from
217.6.190.4
SIP/217.6.190.4:5060-666d is
circuit-busy
Isabove
At the request of a number of community members, I have created a new
mailing list for the purposes of discussing internationalization issues
related to Asterisk (code, configuration, usage, etc.).
The list is called asterisk-i18n, and is available via lists.digium.com
just like all the other
Title: How to do Call Forwarding
Hi all. I am attempting to setup a dial plan which will allow me to forward an extension using the handset. I have followed the instructions in http://www.voip-info.org/wiki/index.php?page=Asterisk%20call%20forwarding however it does not work correctly. Does
What are you plugging into the FXO port? I have two SPA-2000's. One
is connected to an SBC POTS line and the other to a Vonage Cisco
ATA186. The SPA3k connected to the SBC POTS passes CID info, the one
connected to the Cisco ATA will not. I have tried many things
mentioned at AAH SF.net
Hello,
I added these rules to my firewall, and it works fine:
# voip mangle
$IPTABLES -t mangle -A FORWARD -p udp --dport 5060:5069 -j TOS --set-tos
Minimize-Delay
$IPTABLES -t mangle -A FORWARD -p tcp --dport 5060:5069 -j TOS --set-tos
Minimize-Delay
$IPTABLES -t mangle -A FORWARD -p udp
i have implemented something using mysql.. and thus i have a
phone-features page that allows me to login / authenticate using the
voicemail-users table for the pin and extension.. and then set the
destination number, and then turn it on or off.
then in the dialplan, mysql kicks in and checks to
I'm running a T1 between Asterisk and a Nortel Meridian Option 61c.
When I call from Asterisk to the Nortel system, it displays both the
name and number of my SIP phone. When I call from the Nortel system to
Asterisk, I only get CallerID number. Set-to-set calls on the Nortel
system display both
I'm finishing up a first version of a web interface for end
users. It's focus is specific for our own uses, but I plan on
releasing it under an open source license and would appreciate any
feedback while I wrap up the first version.
The interface is designed for end users without any real
I was hoping there would be something considerably more simple.
For example, on my legacy PBX, all I need do is press the Call Fwd
button on my phone, followed by an extension. Something similar (like
*72#ext) would be nice.
David A. Morrow
Technical Systems Lead
Autodata Solutions Company
Kerry Garrison wrote:
A phone plugged into it will grab the CID on about the second ring and I
have adjusted the SPA3000 out to 5 rings with no difference. What gets
passed to asterisk is whatever is set in the 3000's Display Name field. If
the Display Name field is blank, then nothing comes
On Wednesday 26 October 2005 14:03, Dave Morrow wrote:
I was hoping there would be something considerably more simple.
So make it simple. :-)
For example, on my legacy PBX, all I need do is press the Call Fwd
button on my phone, followed by an extension. Something similar (like
*72#ext)
yes.. that is what i was talkinga bout.. it's *xx keysequence followed by
number.. where you dial it from the extension you are trying to forward.
./Ben
I was hoping there would be something considerably more simple.
For example, on my legacy PBX, all I need do is press the Call Fwd
button
What does your extensions.conf look like now to try and implement this? Some prior examples that were based on jumping priorities may not work so well with a stock 1.2b install where priorityjumping=no in extensions.conf.
On 10/26/05, Ben Higley [EMAIL PROTECTED] wrote:
yes.. that is what i was
- Original Message -
From: Chris Shucksmith [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, October 26, 2005 2:17 PM
Subject: [Asterisk-Users] UK BT IDSN30e 'pass through' with
TE205P/AvayaArgentOffice?
Hi all,
I have a shed load of UK (BT) analogue
Hi,
Using a FC4 system without a digium card to centralize a few other systems
with digium cards makes sound to break when zaptel module is loaded. Even
if ztdummy is present.
If i unload zaptel module, it works fine, but when i try to do a call with
trunking, asterisk does not even answer.
On
On 10/26/05, Andy Vega [EMAIL PROTECTED] wrote:
I'm running a T1 between Asterisk and a Nortel Meridian Option 61c.
When I call from Asterisk to the Nortel system, it displays both the
name and number of my SIP phone. When I call from the Nortel system to
Asterisk, I only get CallerID number.
Thanks for the great
feedback! We now have an updated fax2mail (version 2.0) for download (at
www.generationd.com). Fixes
include:
1. Improved
detection of the number of fax pages
2. Handling of
"!" character in the name (for those users ofthe "hoodaheck" module)
3.
Correction of
Common requests from my customers include;
-MACs (moves,adds,changes) on extensions (sip, zaptel,CID)
-Voice Prompt recording/modifying
-CDR Access on the fly
-Reboot/halt option
-The Multi-tenant functionality would be very nice also.Big market for
that.
Hope this helps. Good luck!
On Wed,
This is a DIVA Server card correct? Regular diva or diva pro will not
work as far as I know.
Regards,
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, October 26, 2005 4:31 AM
To: asterisk-users@lists.digium.com
This is a portion of my extensions..
i have also in my mysql db extensions table a
exten = _*23.,1,SetVar(CF_DEST=${EXTEN:3} )
exten = _*23.,2,Noop(Call forwarding to ${CF_DEST} for exten:
${CALLERIDNUM})
then write that value into the database, for that extension that is
calling,
A good solution is use a program that use sipsack for SIP, something like
sipsack for IAX and Linux-HA for asterisk. In this way you check if SIP or
IAX is OK, and if these technologies are bad, you can kill asterisk and
linux-HA will do the rest. In PSTN Field, you can check rxhooksig in struct
Dear Friends,
Great day for the callingcard-lovers !!!
I am pleased to release the version 3.0 of AreskiCC !!!
http://www.areski.net/a2billing/
http://www.voip-info.org/wiki/view/A2Billing
Little unexpected change, we got a new name... bit more serious A2Billing
Many many features have been
Alex Lopez (username 'opsys') has joined our bug marshal team and will
be helping to keep bugs moving through the system and get them
tested/confirmed, so please welcome him and give him lots of work to do!
___
--Bandwidth and Colocation sponsored by
Yes I do have that set to Yes.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Daragon
Sent: Wednesday, October 26, 2005 11:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SPA3000 as trunk - no caller
I am connected to Cox Cable telephone service which nicely passes CID when
using an X100P card but am not getting CID when using the SPA3000. I am
using 3.1.5(GWb) -Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent:
Areski,
The featurelist in this version is Awsome. This will clearly and
absolutely make all those Closed Source Billing systems and so called
Soft Switches like Bicom's Switchware, obsolete.
Kudos for your effort and contribution to the Asterisk users.
This probaly is one of the most
Chris:
ARI has been recently expanded into this space for end user
configuration. Don't know if you looked at it.
http://www.littlejohnconsulting.com/?q=node/11
It is works well coupled with AMP, but can be run stand alone as well.
Installation very easy and there are few dependencies. Just
Hi list, i'm having a problem with asterisk+pstn termination, i just
bought a TDM400p and connect my phone line(bellsouth) now when im using
the pstn through asterisk there's an echo, i don't know if this is
already have been resolved. If it does please point me to the
instruction how to resolve
On 10/26/05, Dan Littlejohn [EMAIL PROTECTED] wrote:
Chris:ARI has been recently expanded into this space for end userconfiguration.Don't know if you looked at it.http://www.littlejohnconsulting.com/?q=node/11
Thanks for posting this. I hadn't seen it up until now.
Chris
I have a small test system -- 6 phones. It is a dual processor server. I
noticed that asterisk spawns 12 child processes. Can this be controlled? I
would think 2-4 would be plenty for this test site.
Thanks
--john
--
This mail was scanned by AntiVir Milter.
This product is licensed for
Just wondering if anyone has any example applications that
they are using with the Polycom microbrowser? I have done some simple hello
world applications and such but would like to add call parking and didnt
want to waste time re-inventing the wheel.
Also, does anyone know if you can
Did you ever get those files translated?
begin:vcard
fn:Jason Garland
n:Garland;Jason
org:Zentality
adr:;;1000 E. 116th St.;Carmel;IN;46032;USA
email;internet:[EMAIL PROTECTED]
title:Systems Engineer
tel;work:1-317-818-6956
x-mozilla-html:FALSE
url:http://www.zentality.com
version:2.1
end:vcard
On Oct 26, 2005, at 3:40 PM, Mark Quitoriano wrote:
Hi list, i'm having a problem with asterisk+pstn termination, i just
bought a TDM400p and connect my phone line(bellsouth) now when im
using the pstn through asterisk there's an echo, i don't know if this
is already have been resolved. If it
Hi list, i'm having a problem with asterisk+pstn termination, i just bought
a TDM400p and connect my phone line(bellsouth) now when im using the pstn
through asterisk there's an echo, i don't know if this is already have
been resolved. If it does please point me to the instruction how to
Kerry Garrison wrote:
Yes I do have that set to Yes.
Does the SPA-3000 show the caller ID in the last call field in the
summary page ? It's capable of interpreting a bewildering array of
callerid schemes - is it set to what your local telco is generating ?
jd
--
John Daragon
During a PSTN call the status screen correctly displays the caller ID
information.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Daragon
Sent: Wednesday, October 26, 2005 2:41 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Hi there,
This is incredible - we've been running an Option 11C with Asterisk for
months and have never found anyone else doing the same and now it's
becoming a weekly occurrence!
We have exactly the same issue and have not been able to find an answer
for it. What we did as a workaround (we
[EMAIL PROTECTED] wrote:
Areski,
The featurelist in this version is Awsome. This will clearly and
absolutely make all those Closed Source Billing systems and so called
Soft Switches like Bicom's Switchware, obsolete.
Seshu,
We welcome competition of any kind. It just makes us improve and
On Oct 25, 2005, at 6:33 PM, Jerry Richmond wrote:
This means I won't be giving anyone my personal
information.
I haven't laughed so hard in ages.
lyd
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Sean Cook wrote:
Just wondering if anyone has any example applications that they are
using with the Polycom microbrowser? I have done some simple hello
world applications and such but would like to add call parking and
didn’t want to waste time re-inventing the wheel.
Also, does anyone know
On 10/26/05, Chris HARIGA [EMAIL PROTECTED] wrote:
I have a show parked calls php script for my Polycom IP600 phones. If
U are interested let know and I can email it.
Even if Sean doesn't want it, I do! All examples can be helpful. :-)
Why not put up a page on the wiki linked from the polycom
Gary Reuter wrote:
On 10/26/05, *Chris HARIGA* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I have a show parked calls php script for my Polycom IP600
phones. If
U are interested let know and I can email it.
Even if Sean doesn't want it, I do! All examples can be
Hello All , I installed a x100p clone device in a system .
I pulled (Wed, 26 Oct 2005 early am) cvs compiled installed for a
2.6 kernel . We call this system 'test' . This system is a HP Vectra
with a 933MHZ processor . Test has all the z* modules installed
Thought this may be of interest to some people on this list.
https://studio.tellme.com/skype/submissionprocess.html
Shame we were never able to get Tellme to get their act
together to for an Asterisk gateway as per http://www.voip-info.org/wiki/view/Tellme
Regards,
Dean
First I don't like the 6 line cord. Use an rj11 2 wire cord, but watch the
crossover vrs straight on the old red and green.
Next the interrupt must be fixed. Do this in the CMOS before you boot. Go
to the PCI bus assignments and set the IRQ or go and disable the serial
ports thereby allowing
Armin, thanks for your response
My problem now is that having the configuration on capi.conf as shown in my
original post I am not able to receive incoming neither make outgoing calls.
When making an outgoing call I get the following debug and verbose output
from asterisk,
-- Goto
Hi all,
Recently, I have builded a small call center for our company. Till now, But they want it auto-sayagent's work number once the agent pick up the call.
Any one knowhow to do that?
Any help and advice will be appreciated!
Thanks a lot!
Best Regards,
Gary Li
雅虎免费G邮箱－中国第一绝无垃圾邮件骚扰超大邮箱
Will this work if I am using text file configs?
I started with AMP, but didn't like the limitations.
I disabled the DB config parts, but still use the other features of AMP.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
----
Hi All,
With asterisk and call manager hooked up via the sip trunk,
the calls from ccm and asterisk can call each other. I have 2 problems.
Is it possible to route all
calls via the call manager and not via asterisk when I dial any number?
This is divided into 2 problems
Steven:
I believe so. If you recieve database errors you will have to disable
those parts of ARI as well, but it might run out of the box without
problems.
Dan
On 10/26/05, Steven [EMAIL PROTECTED] wrote:
Will this work if I am using text file configs?
I started with AMP, but didn't like
I have two TE110P cards.
If I stop the Zaptel service, the whole server hangs.
I have had this issue with 1.0.7, 1.0.8 ,1.0.9 and 1.0.9.2.
The server is a Dell 1750 with all unnecessary BIOS options off (USB,
Serial, Second NIC, etc)
It is Dual CPU.
There are no shared Interrupts.
please advise
Michael J. Lynch wrote:
Has anyone else noticed that make install for
asterisk-sounds-1.2.0-beta1 fails. The problem is that the last 10
or so entries in file sounds-extra.txt (starting at line 2119) dont have
the .gsm extension on the file name. E.G.
Yes, this has already been fixed in CVS.
With asterisk and call manager hooked up via the sip trunk, the calls
from ccm and asterisk can call each other. I have 2 problems.
1. Is it possible to route all calls via the call manager and not
via asterisk when I dial any number?
Yes
1. This is divided into 2
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