I'm taking the liberty to announce this event on the Asterisk mailing
list, as Asterisk and OpenSER form a valuable combination in SIP
architectures.
The second edition of OpenSER Summit will take place in San Jose, USA
,on the 17th of March, 2008, during VonX Spring 2008 pre-conference
events.
Has anyone here any experience in getting an Asterisk box to talk to
a Cisco Unity system? I have a potential customer who would like to
add a conference bridge to their existing Cisco Unity setup.
The digging I have done so far suggests that it should be possible to
talk SIP between them, but
Leap year? Election year? Will your GoToIfTime() dialplan function
properly on Feb 29th?
Every week we try to get guests with ideas, products and services you
haven't had time to check out to come and talk about what they're
doing. Aastra has some interesting phones so we asked them to come
talk
hello everyone
what i did to configure SPA3102 is
-sip.conf-
;spa-fxs
[108]
type=friend
host=dynamic
context=sipphones
secret=VerySecretPass
mailbox=108
dtmfmode=rfc2833
;dtmfmode=inband
disallow=all
allow=alaw
;spa-fxo-in
[118]
type=friend
Hi All;
We have a telephony service provider that is asking
what is new technology and services to be added with
the telephony service that can be used for VoIP and
PBX purposes.
Any suggestion to be added that can really give new
advantages and technologies specially in VoIP issues?
Anyone
Do you mean Call Manager? Unity is just their voicemail system. Yes,
you can use SIP to talk between * and CM. You can also use h.323, but
it is a big hassle.
Tony Mountifield wrote:
Has anyone here any experience in getting an Asterisk box to talk to
a Cisco Unity system? I have a
im trying to use monitor() aplication with b option, to start the
recordigin just once the conversation has actuallly begun.
It works fine with a sip extensión, but when i use a zap channel, it records
all the channel bridging, including the ringing sounds...
could you please help me with this
register = [EMAIL PROTECTED]:X:[EMAIL PROTECTED]
[8159093010]
fromdomain=proxy.essex1.com
host=proxy.essex1.com
port=5060
insecure=very
username=8159093010
secret=X
type=peer
qualify=no
canreinvite=no
dtmfmode=rfc2833
disallow=all
allow=ulaw
outboundproxy=proxy.essex1.com
[Feb 28
On Thursday 28 February 2008 05:13:06 randulo wrote:
Will your GoToIfTime() dialplan function properly on Feb 29th?
It will work fine. In fact, you can put February 30th or February 31st into
your GotoIfTime arguments, and it will accept the values just fine (it just
won't ever evaluate true).
Do you have an English translation of this post?
On Thu, Feb 28, 2008 at 6:48 AM, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi All;
We have a telephony service provider that is asking
what is new technology and services to be added with
the telephony service that can be used for VoIP and
In the telephony world, this is called glare, it's most prevalent on
Analog (though you can have the same thing happen with robbed-bit
T1). There really isn't much you can do to prevent it, only minimize
it. You need to have your inbound and outbound starting at opposite
ends. If your
On Thu, Feb 28, 2008 at 08:47:37AM -0600, James Texter III wrote:
On Feb 27, 2008, at 2:55 PM, Tim Nelson wrote:
Hello! I've run into a problem where a user is making an outbound
call at the same time that an inbound call is being made on the same
analog line. It appears that as the zap
Thank you all for the suggestions. I'm looking into getting groundstart lines
for that installation as suggested earlier. Also, I'll try setting the outbound
call routes in reverse from the inbound hunt group. I appreciate your help!
Tim Nelson
Systems/Network Support
Rockbochs Inc.
-
On Thu, Feb 28, 2008 at 3:16 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Thursday 28 February 2008 05:13:06 randulo wrote:
Will your GoToIfTime() dialplan function properly on Feb 29th?
It will work fine. In fact, you can put February 30th or February 31st into
your GotoIfTime
:-) HAHA.. Unfortunately, PRI service is not available at this location...
Thank you for the help!
Tim Nelson
Systems/Network Support
Rockbochs Inc.
- Original Message -
From: Jay R. Ashworth [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, February 28, 2008
Dear all,
I've created a digium certified asterisk professional - dCAP linkedin
group for anyone, dCAP, interested:
http://www.linkedin.com/e/gis/60298/39AE1350DBF3
Best regards,
Marco Mouta
dCAP
November 2006
--
Esta mensagem (incluindo quaisquer anexos) pode conter informação
confidencial
This may fix your issue:
mx,10,4,0
By default Mexico variant has the option get ANI after DNIS. Which
it means just after getting the DNIS digits we will request the
calling party category and DNIS. The Nortel PBX seems to not like
calling party category requests and they want to go straight to
Hi, I believe your problem of authorization is relative to astersik's
manager.conf configuration and you need to add and user and password in the
manager.conf to have remote access. I have used some examples of voip-info.org,
look at this link in the second half part, it explain how to
I'm pretty sure he's asking what sort of advantages there are in using
VoIP (and probably Asterisk) over traditional wireline services.
Advantages being things like flexibility and portability (with cost and
barriers-to-entry being somewhat debatable). But he's more interested
perhaps in the
Hi,
I am trying to build zaptel 1.4.8 with kernel 2.6.24 on debian/sid:
zenon:~# module-assistant -t build zaptel
make[3]: Entering directory `/usr/src/linux-2.6.24.3'
scripts/Makefile.build:46: *** CFLAGS was changed in
/usr/src/modules/zaptel/Makefile. Fix it to use
Raul Alarcon wrote:
im trying to use monitor() aplication with b option, to start the
recordigin just once the conversation has actuallly begun.
It works fine with a sip extensión, but when i use a zap channel, it
records all the channel bridging, including the ringing sounds...
could
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Same effect...
I belive that is a nortel issue.
But I have no idea of how to debug it to fix it... any advice is helped..
Also the provider, asked me for the tone table because he can set
the tone table as he wishes...
TIA.
Testcalll output
Tony wrote:
Has anyone here any experience in getting an Asterisk
box to talk to a Cisco Unity system? I have a
potential customer who would like to add a conference
bridge to their existing Cisco Unity setup.
The digging I have done so far suggests that it should
be possible to talk SIP
Louis-David Mitterrand wrote:
zenon:~# module-assistant -t build zaptel
make[3]: Entering directory `/usr/src/linux-2.6.24.3'
scripts/Makefile.build:46: *** CFLAGS was changed in
/usr/src/modules/zaptel/Makefile. Fix it to use EXTRA_CFLAGS. Stop.
Is there a quickfix out
Subject: I Would Rather See People Running Asterisk
on a Real Server than an Emachine
It's funny you say that - I've got a 667MHz eMachine with 256mb of RAM
running Trixbox + hylafax/iaxmodem, routing our Internet traffic
w/iptables, proxying the kids' net traffic w/squid, samba...
Hi all,
I've just got an OpenVox A400P card with 1 FXO and 1 FXS module and I am just
trying to get it working. But no luck as of yet.
In /etc/zaptel.conf I've set the following options:
fxsks=2
fxoks=1
loadzone=se
defaultzone=se
And in /etc/asterisk/zapata.conf I've not sure what to set exactly.
Hi;
Yes what u said is correct, I am interested in using
VoIP and Asterisk (also) over wireline services
(telephon line). Actually the service provider company
asking for such things to be added with the telephone
lines that they give it for their customer. Actually
they build 9 PSTN in the
Thanks for the info, Dan Peder. It helps me to know the right questions
to ask the customer!
Cheers
Tony
In article [EMAIL PROTECTED],
Dan Austin [EMAIL PROTECTED] wrote:
Tony wrote:
Has anyone here any experience in getting an Asterisk
box to talk to a Cisco Unity system? I have a
arkda wrote:
Nothing in the console aside from what I've posted. When a DTMF tone is
played the server freezes instantly, hard reboot required.
Just to close out this thread, it appears that this issue was related to
http://bugs.digium.com/view.php?id=12053
Adding a loadzone and
Hello , about this implementacion , i have a issue with ASterisk 1.4.2 and
Cisco Unity , the VM doesn't work fine the calls are good but when enter the VM
( cisco Unity ) it didn't work .
Somebody has one implementacion ?
To: asterisk-users@lists.digium.com From: [EMAIL PROTECTED]
I think Bilal's service provider is asking What is the next Killer Ap for
VoIP?
--Don
Don Kelly
PCF Corp
Real Support for your Virtual Office TM
651 842-1000
888 Don Kell(y)
651 842-1001 fax
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bilal
Tim Nelson wrote:
Thank you all for the suggestions. I'm looking into getting groundstart lines
for that installation as suggested earlier.
Make sure your interface supports GS
The Sangoma and TDM cards do
I assume you are using one of these as you mention Zaptel.
John Novack
Also, I'll
Yes... this installation has a Sangoma A400D card fully populated. Thanks again.
Tim Nelson
Systems/Network Support
Rockbochs Inc.
- Original Message -
From: John Novack [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
I understand the use of the g option in a call of Set() is deprecated
as of version 1.4.
Was the GLOBAL function used to replace it introduced in version 1.4 or
were there some late 1.2 versions that also supported it?
--
Phil Reynolds
o mail: [EMAIL PROTECTED]
|L_ \ / Web:
Heres a little teaser for those of you with iPhones
Asterisk Voicemail for iPhone allows you to check your voicemail
messages on your house or business line from your iPhone. You can
think of it as Visual Voicemail, but for your Asterisk PBX numbers
instead of your ATT cell number. The technology
On Thursday 28 February 2008 16:04:49 Phil Reynolds wrote:
I understand the use of the g option in a call of Set() is deprecated
as of version 1.4.
Was the GLOBAL function used to replace it introduced in version 1.4 or
were there some late 1.2 versions that also supported it?
It was
Anthony Messina wrote:
Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing
out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3.
I get the following:
This should be fixed in Zaptel 1.4.9.2.
--
Russell Bryant
Senior Software Engineer
Open
This looks great, Cant wait to try it on my iphone
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Carey
Sent: Friday, 29 February 2008 9:18 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk Voicemail for iPhone
Heres a
Hi all,
Using the latest test version of Debian but when I have done modprobe -r and
removed a few of the zaptel modules some of them cannot be removed. The other
module is in use. Also if I reboot my system they're all loaded again. Any
thoughts?
Many thanks,
Christian
Is this only on the _64 zaptel or will affect ALL zpatel 1.4.9 ?
-Original Message-
From: Russell Bryant [EMAIL PROTECTED]
Sent: Feb 28, 2008 6:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] TDM400P dialout
On Thursday 28 February 2008 05:41:55 pm Al Baker wrote:
Is this only on the _64 zaptel or will affect ALL zpatel 1.4.9 ?
-Original Message-
From: Russell Bryant [EMAIL PROTECTED]
Sent: Feb 28, 2008 6:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
/etc/modprobe/blacklistor similar
PaulH
On Fri, 2008-02-29 at 00:30 +0100, Christian wrote:
Hi all,
Using the latest test version of Debian but when I have done modprobe -r and
removed a few of the zaptel modules some of them cannot be removed. The other
module is in use. Also if
Anthony Messina wrote:
i'm looking forward to 1.4.9.2, but am also concerned about
http://bugs.digium.com/view.php?id=12099 as i saw this error with 1.4.9 and
1.4.9.1 on both platforms.
kpfleming has done some work today on this issue which needs a little
more in house testing.
In the
If anyone has managed to compile and run Asterisk on a server from this
particular era, I'd /love/ to know about it. :)
What's the performance like? For that matter, what phones were
available at the time?
randulo wrote:
On Thursday 28 February 2008 05:13:06 randulo wrote:
Will your
Hi All,
If i have this kind of setup, what do i need to make it's load balance.
[ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ]
| | | |
-
|
Dean Collins will sell you ideas.
On Thu, Feb 28, 2008 at 4:32 PM, Don Kelly [EMAIL PROTECTED] wrote:
I think Bilal's service provider is asking What is the next Killer Ap for
VoIP?
--Don
Don Kelly
PCF Corp
Real Support for your Virtual Office TM
651 842-1000
888 Don Kell(y)
On Fri, Feb 29, 2008 at 2:01 AM, Ron [EMAIL PROTECTED] wrote:
Hi All,
If i have this kind of setup, what do i need to make it's load balance.
[ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ]
| | | |
Yes, try http://1ezphone.com its a browser softphone.
- Original Message -
From: Zoa
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP Softphones and Citrix ?
Date: Fri, 01 Feb 2008 23:09:56 +0200
I'm working for zoiper.com and
I would like to hire someone to automate my asterisk for hosted PBX
service for fetures like user signup, adding money and call bridging
Please contact me offline at [EMAIL PROTECTED]
- Original Message -
From: Philipp Kempgen
To: Asterisk Users
Subject: Re: [asterisk-users]
Hi Greyman,
Should it look like this now? Can i use 2 SIP Proxies just to make sure
i have a backup? will that cause any problem again with regards to
calling extension to extension? Extensions will register on the asterisk
still? How about outbound calls to other SIP provider or a
If i have this kind of setup, what do i need to make it's load balance.
[ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ]
| | | |
-
|
On Fri, Feb 29, 2008 at 4:03 AM, Ron [EMAIL PROTECTED] wrote:
Hi Greyman,
Should it look like this now? Can i use 2 SIP Proxies just to make sure
i have a backup? will that cause any problem again with regards to
calling extension to extension? Extensions will register on the asterisk
On Fri, 29 Feb 2008 6:21 +0200, Yehavi Bourvine +972-8-9489444
[EMAIL PROTECTED] wrote:
See the discussion a few days ago. The Asterisk server saves the value of
SYSNAME (defined in asterisk.conf) in the field REGSERVER inside MySQL.
Regards, __Yehavi:
Ahh
Hi all,
I have been googling and testing without any luck and would appreciate
any guidance from anyone.
A port has already been configured on the CISCO switch with the
following:
interface FastEthernet2/0/1
description VOIP VLAN 100
switchport access vlan 100
switchport mode access
duplex full
You can paste and copy nterface FastEthernet2/0/1 switchport access vlan
20 switchport mode access switchport voice vlan 120 srr-queue bandwidth
share 10 10 60 20 srr-queue bandwidth shape 10 0 0 0 mls qos trust
device cisco-phone mls qos trust cos auto qos voip cisco-phone
spanning-tree
On Fri, Feb 29, 2008 at 12:30:49AM +0100, Christian wrote:
Hi all,
Using the latest test version of Debian but when I have done modprobe -r and
removed a few of the zaptel modules some of them cannot be removed. The other
module is in use. Also if I reboot my system they're all loaded again.
On Thu, Feb 28, 2008 at 08:08:49PM +0100, Christian wrote:
Hi all,
I've just got an OpenVox A400P card with 1 FXO and 1 FXS module and I am just
trying to get it working. But no luck as of yet.
In /etc/zaptel.conf I've set the following options:
fxsks=2
fxoks=1
loadzone=se
defaultzone=se
Thanks very much for the quick response.
However, switchport voice vlan.. I thought is only valid for CISCO phones
and I am using Polycom and thus it would not work.
Furthermore, I have already tried switchport voice vlan... before I emailed
to the list.
I encountered this strange problem which is I can call into Asterisk box
but I cannot call out.
I was successful before using exactly the same euroISDN line but with
TE110 and different versions of Asterisk.
This time, I am using:
. TE410
. Asterisk 1.4.13
. Zaptel 1.4.6
. Libpri 1.4.2
1) I put
59 matches
Mail list logo
