Hi Steef,
Do you want to send me an email to [EMAIL PROTECTED] and I can assist you
further.
It should work as far as I am aware.
Thanks David
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of asterisk
Sent: 28 November 2005 07:51
To: Asterisk Users Mailing
Original Message
From: Innocent Evil [EMAIL PROTECTED]
you can use 'w' option with 'Dial' on 1.2.x
I don't think w do anything like 'wait', If I am wrong, correct me
someone please According to app_dial.c
w- Allow the called party to enable recording of the call by
This is the purpose of AstBill at http://astbill.com
It gives you cost based CDR's out of the box.
AstBill is an Open Source Web Based Billing, Routing and Management
Software for Asterisk and VOIP. AstBill Provides pre and post paid
billing services and have a calling card module. AstBill
HI,
I have been playing around with the A2billing agi for a few days( I
really think its a great application) and trying to understand its
working. As far as I have understood, the only debugging mechanism is
going through the logs it generates.
It would be really cool if I can run the agi in
Vedran Dakic ha scritto:
How does Asterisk handle this kind of setup with one-two/cluster
central server(s) and a bunch of other servers
connected with IAX(2)? If you have local calls, do they go directly
from phone to phone, do they go from phone to
per-floor-Asterisk server, or
On 11/25/05, Daniel Wright [EMAIL PROTECTED] wrote:
Steve Davies wrote:
Hi,
This is probably just me mis-reading the documentation, but I have
been led to believe that the '.' in extensions.conf means zero or more
digits, such that
exten = _X.,1,NoOp()
Would trigger for either a
On Sun, Nov 27, 2005 at 05:22:39PM -0300, Rodrigo Campos wrote:
On 11/27/05, Geotrix [EMAIL PROTECTED] wrote:
Hello,
I am trying to install zaptel wcfxo with X101.P board on Debian sarge
without success.
(previously compiled and worked OK on Redhat kernel)
and in debian ?
for
On Sun, Nov 27, 2005 at 07:40:04PM -0800, Geo wrote:
Well, thanks, it might be great your package yet I would like to know how to
adapt.
I wouldn't like to rewrite Debian neither Asterisk but is somebody able to
advice
how you define modules in zconfig.h or whatever ?
Any tip ?
Geo
This is so simple, you are going to kick yourself. Type agi debug in
the console.
Thanks,
Steve
-Original Message-
From: Danish Samad [mailto:[EMAIL PROTECTED]
Sent: Monday, November 28, 2005 3:52 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] a2billing / php agi
Dear Tzafrir Cohen,
I paste the detail message for you,please see the below message.
I wonder it may cause by different linux system.
[EMAIL PROTECTED] speech-coding]# patch -p1 /tmp/ipp-050903.diff
patching file G723.1/Makefile
patching file G723.1/samples/codec_g723.c
patching file
I've looked through the archives of the mailing list for the
last year and although informative I've not been successful
at get this to work. We had a working Asterisk PBX system
with 3 Digium X101P FXO lines and two TDM400P FXS cards.
I've setup an ADIT 600 with an 8 port FXO card (and an
8 port
HI Steve,
Thanks for the reply.
I have tried agi debug before. Actually by debugging I mean
inserting breakpoints and stepping through the code. I have not been
able to find any solution for this sort of debugging setup.
Regards,
DanishOn 11/28/05, Steve Totaro [EMAIL PROTECTED] wrote:
This is
Hi,
We are trying to integrate Asterisk in front of our existing legacy PBX:
outside line -- FXO -- Asterisk PC -- FXS -- PBX -- Extensions
Asterisk answers outside calls and the IVR asks the user to dial extension #.
The problem is, that when Asterisk forwards calls from the outside line
to the
Hi,
We are trying to integrate Asterisk in front of our existing legacy PBX:
outside line -- FXO -- Asterisk PC -- FXS -- PBX -- Extensions
Asterisk answers outside calls and the IVR asks the user to dial extension #.
The problem is, that when Asterisk forwards calls from the outside line
to
Hi, anyone managed to get a Presence
Agent configuration with Asterisk 1.2 and X-Ten Eyebeam working. I believe
this should be paritally supported now in 1.2 ?
regards
Mark___
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Asterisk-Users
Hi,
We are trying to integrate Asterisk in front of our existing legacy PBX:
outside line -- FXO -- Asterisk PC -- FXS -- PBX -- Extensions
Asterisk answers outside calls and the IVR asks the user to dial extension
#.
The problem is, that when Asterisk forwards calls from the outside line
Vedran,
Email me off topic and I can provide you some case studies of different
providers for your review.
[EMAIL PROTECTED]
Roger Workman
Business Development
Upperclassman/Universal Holdings LLC
Voice: 304.324.3800
Fax: 304.324.3801
ICQ: 4447584
Website: http://www.upperclassman.net
Don't waste your time asterisk does not support
presence
--- Mark van Kerkwyk [EMAIL PROTECTED] a écrit :
Hi, anyone managed to get a Presence Agent
configuration with Asterisk 1.2
and X-Ten Eyebeam working. I believe this should be
paritally supported
now in 1.2 ?
regards
Mark
Hi
it's possible to upgrade the firmware of a cisco 7910 with asterisk ?
he have a other solution for upgrade it without callmanager ?
thansk for your help
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To
Are you sure? I've got it working with Eyebeam, showing me just who is
available and who isn't.
http://www.voip-info.org/wiki-Asterisk+phone+snom
A couple of pages down you'll see this:
SNOM SUBSCRIBE/NOTIFY support for monitoring extension states
The methods and configuration here are
Noc Phibee ha scritto:
it's possible to upgrade the firmware of a cisco 7910 with asterisk ?
You need the legal firmware upgrade file
download the chan_sccp code from http://chan-sccp.berlios.de
configure it and use the imageversion param to upgradde the phone firmware.
Of course you need a
I'm sure look at rfc3265 (SUBSCRIBE/NOTIFY) which is
not support by asterisk.
How can you monitor the states of the buddies ?
Harry
--- Ben Buxton [EMAIL PROTECTED] a écrit :
Are you sure? I've got it working with Eyebeam,
showing me just who is
available and who isn't.
aha, it was the subscribecontext= that
was missing. Basic presence works fine now. Offline, online, on the phone
:-)
What about IM, did you have that working
too ?
thanks
Mark
Ben Buxton [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
28/11/2005 11:59 PM
Please respond to
Asterisk Users
Can't say I've actually tried IM, but Ill give it a go sometime. I think
the wiki needs updating on all this...the eyebeam page is very
incomplete on subscribe, im, etc.
BB
Mark van Kerkwyk [EMAIL PROTECTED] uttered the following thing:
aha, it was the subscribecontext= that was missing. Basic
hi list!
I've configured some voicemailboxes and at the beginning everything was
working fine. In the past few days, evetime i want to hear the messages,
recorded on my box the following lines come up the asterisk logfile:
Nov 28 14:12:31 WARNING[25446]: file.c:508 ast_openstream_full: File
Thank you for your asnswer
I found that between 7 and 8 in the mornig I have a low load of my box.
I modify my script in this way:
asterisk02:/ # cat /closeasteriskandreboot.sh
#!/bin/bash
echo chiusura schedulata when convenient di asterisk
/usr/sbin/asterisk -rx stop when convenient
hat is the content of your wanpipe?.conf files ?On 11/26/05, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Stig Even Larsen wrote: I'm having problems connecting my Sangoma cards to our PRI (E1)
interface. It seems that the card get connected (green led), but Asterisk reports: Status: Provisioned,
I looked into why I can't get the original DID number called when a fax
is detected (so I can later route to the correct email address). There
is a variable called FAXEXTEN that is created when a fax is detected,
but it is not being populated with the original extension / did number
called. It
I have a problem when i have asterisk connected to voxtream parlay i60
PRA port.
When call is received to asterisk and forwarded to SIP IP gsm gateway
call is always disconnected with cause 102.
# 102 Recovery on timer expiry
This cause indicates that a procedure has been initiated by the expiry
Hi Stig,
When I install another Sangoma card on the same system (pri_net), and
connect the two cards with a PRI cross-over cable both cards get connected
(green led) and Asterisk reports both spans:
Status: Provisioned, UP, Active
It does look your installation is fine, i.e you are being able to
Hello list,
this is just an announce of a new mailing list dedicated to deploying,
running and managing real-
world Asterisk-based call centers. The mailing list is in English and
allows knowledge sharing
for this very important - and yet somehow less considered - Asterisk
deployment area.
Found out why there is no original DID set. It looks like while waiting
for the incoming digits timeout (DID), we are getting the fax tone
detect and it is sending a digit 'f' which immediately starts the fax
extension before the incoming DID has been saved.
Is there a way to set in the zap
Hi Guys,
Having a little problem with Realtime Extensions.
I've created the table, (using the same database as i use for realtime peers/users), however when a call comes through, the CLI shows the following warning:-
Nov 28 15:13:08 WARNING[7522]: res_config_mysql.c:623 mysql_reconnect: MySQL
Hello,
I've noticed my AGI, in Perl, was always returning 0 even if exit from
it with something else than 0.
On http://www.voip-info.org/wiki/view/Asterisk+cmd+AGI, it's said :
[AGI] Returns -1 on hangup or if application requested hangup, or 0 on
non-hangup exit. But I tried also to hang up
It should build wcfxo. Not trying anything special.
I just follow the procedure !
When I reboot I have:
ISDN subsystem Rev: 1.1.2.3/1.1.2.3/1.1.2.2/1.1.2.3/1.1.2.2/1.1.2.2 loaded
!!!
HiSax: Linux Driver for passive ISDN cards
HiSax: Version 3.5 (module)
HiSax: Layer1 Revision 2.46.2.5
It should build wcfxo. Not trying anything special.
I just follow the procedure !
When I reboot I have:
ISDN subsystem Rev: 1.1.2.3/1.1.2.3/1.1.2.2/1.1.2.3/1.1.2.2/1.1.2.2 loaded
!!!
HiSax: Linux Driver for passive ISDN cards
HiSax: Version 3.5 (module)
HiSax: Layer1 Revision 2.46.2.5
Does anybody know where I can download ringtones for Cisco 7960's? Need
to be .pcm files.
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Hi,
I understand that a fax machine cannot connect through a Digium TDM400p
card (FXS connected to fax and FXO connected to a pots line) but can
spandsp send and receive faxes as an intermediary between the pots line
and the fax machine.
Thanks
Hi,
I have noticed that most mobile phones (GSM and CDMA at least) seem to
have a tendency to interrupt the incoming audio stream when the
microphone levels get louder than a certain threshold (such as when you
are speaking into it). I do not know exactly why this happens, nor
whether it is
Hi
anyone know if a Trunk SIP howto are created ?
I have 8 VoIP account with for all 1 login/pass per number.
i want add into my asterisk but not know where ;=)
Other questions:
my supplierhave a dns:sip.phonesystems.net
this name have 2 IP address
it's not a problems for Asterisk that he
Thanks sergio for your answer.
But cisco france say me that i cant' bye SmartNet contract on this product.
Only one solution are possible: Bye a special contract at $180.00 ...
Pff i can bye a new equipment with this price hihihi
i can't guest the latest firmware, for me i thinks that the
Hi all,
Does anybody have any info on a decent quality sip hard phone that is
headset compatible?
Thank you
John
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I am trying to trouble shoot some problems with DTMF over PRI. I have a
digium wct1xxp card and these lines in extensions.conf:
exten = 5556000,1,Record(testtone:gsm)
exten = 5556000,2,Wait(2)
exten = 5556000,3,Playback(testtone)
I called in over the PSTN --to-- Asterisk. I did a pri debug,
Noc Phibee wrote:
Thanks sergio for your answer.
But cisco france say me that i cant' bye SmartNet contract on this product.
You can, but only in the US I believe. I've never found any deal less
than £150 (UK).
Only one solution are possible: Bye a special contract at $180.00 ...
Pff i
Noc Phibee ha scritto:
But cisco france say me that i cant' bye SmartNet contract on this
product.
why not?
You can buy those smartnet contract via internet. You just need to mail
US cisco and ask them for the contract activation.
Only one solution are possible: Bye a special contract at
Hello, Rusty!
Thanks for your reply... You have been the only one ;)
Hahaha, it could become dangerous... ;)
Well, I have been investigating it a little bit, and I guess the main
reason is something related to what you have pointed in the first
paragraph.
I've found some interesting information
The only one I can think of to decent price level is the Grandstream GXP
2000. Also have headset jack¨
Anders
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Fraser
Sent: den 28 november 2005 17:27
To: asterisk-users@lists.digium.com
Subject:
Thanks all for your answer ...
all smartnet contrat have access to all firmware in voip ?
thanks
Ryan Amos a écrit :
Cisco phones are not ideal for single-phone setups. If you were to have a lot of them, a $180 support contract is no big deal... However, for Europeans, there should be an $8
Hello all,
Since upgrading a couple of our servers to 1.2 I've noticed problems when
talking to users on 1.0.9 servers. The servers are connected via IAX2 with
trunking and jitter buffer enabled (jitter buffer on default settings).
Reading through posts in the list archives, there are a number
On Monday 28 Nov 2005 15:39, Frank McCarthy wrote:
Does anybody know where I can download ringtones for Cisco 7960's? Need
to be .pcm files.
Download the ringtone generator from Grandstream and make your own.
B
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On Mon, 28 Nov 2005 08:26:57 -0800, John Fraser wrote:
Hi all,
Does anybody have any info on a decent quality sip hard phone that is
headset compatible?
Thank you
John
Aastra 480i ( Ilove it!), Polycom IPx00/x01 series, Snom's all provide
for headsets. Most via RJ style connections, some
hi,while a user talking on, if he gets a new call from queue he hears some noises about this call. I think this happens to inform user about new coming call. But it boring user... So can i stop this noise?Thanks.-ek
Yahoo! Music Unlimited - Access over 1 million songs. Try it
Hi.
I think the best way to do it, is just a IAX2 between the 2 *'s.
Regards.
Juan.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Hiu Yen Onn
Enviado el: Domingo, 27 de Noviembre de 2005 10:50 p.m.
Para: asterisk-users@lists.digium.com
Asunto:
You can, but only in the US I believe. I've never found any
deal less than £150 (UK).
I was quoted £36 a couple of weeks ago by one of the Cisco resellers a
google search provided me with, if that's any help. I can't remember the
company name I'm afraid...
Regards,
Chris
--
C.M. Bagnall,
Hi,
we have following setup : PBX - Parlay -ISDN PRI- Asterisk -SIP- GSM
Gateway
Call comes from PBX through Parlay to Asterisk and it routes it over SIP to
GSM gateway. GSM gateway gives back call progress (it takes some time to
ring or get through), but this info won't get back to Parlay
Depends on the type of headset. The Grandstream GXP-2000 and Liksys SPA-941
have headphone jacks on them, most phones are compatible with Plantronics
univeral headsets.
Kerry Garrison
Publisher - GeekGazette.com - VOIPSpek.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
On Monday 28 Nov 2005 16:29, Tony Hoyle wrote:
Noc Phibee wrote:
Thanks sergio for your answer.
But cisco france say me that i cant' bye SmartNet contract on this
product.
You can, but only in the US I believe. I've never found any deal less
than £150 (UK).
I guess that should depend
What does the TE406 leds indicate?
On 11/28/05, William K. Volkman [EMAIL PROTECTED] wrote:
I've looked through the archives of the mailing list for the
last year and although informative I've not been successful
at get this to work. We had a working Asterisk PBX system
with 3 Digium X101P
According to the Asterisk wiki, adding the delete=yes option to a
voicemail definition should automatically delete messages after they
are emailed. This is the format that I'm using:
101 = ,First Last,[EMAIL PROTECTED],attach=yes|delete=yes
When I leave a message at mailbox 101, it gets
I have been trying to work this problem out with my IAX provider.
I dial a toll-free number, ex: 1-888-876-6262, and I get a due to technical difficulties message.
I set my debug level to 9, and all I see when I dial out is this:
-- Executing Dial(SIP/27-51de, IAX2/voctel/1766262||T) in new
We just upgraded to Asterisk 1.2 a few days ago. And now the Retrieve
voicemail and hold buttons on our SNOM 360 phones are no longer
working. When you put a caller from one of our zaptel lines on hold it
hangs up on them immediately. Interestingly, if you put an internal
extension on hold it
I apologize for the resend. I haven't received much feedback from this.
I also noticed that what I'm getting is the caller id as the caller
name and the sip peer name as the caller id number.
Does anyone have any ideas/suggestions?
Thanks,
Waldo
On Nov 26, 2005, at 2:52 AM, Waldo
The Asterisk development team is pleased to announce that we have
migrated our project repositories and development processes over to the
Subversion version control system!
Effective immediately, the primary source code distribution point for
Asterisk, Zaptel and other related projects (other
I'm using asteriskathome 2.0 beta 6 (asterisk 1.2) with SIP softphones,
7905G cisco SCCP and analog phone( DTMF dialing). All is working nice,
however when I change DTMF for an analog pulse dialing,my analog phone
is not working.
I've found the following :
I haven't heard of this product before so I did some searches on the Internet, this card is $5,400 for a single span T1 card? ouch!
http://www.eiconworks.com/DivaServerV-PRI_T1%20.asp
On 11/25/05, David Waugh [EMAIL PROTECTED] wrote:
Hi John,I'm going to have to disagree with some previous
Hi,
I have an AGI script that is called after receving a call on a channel.
And my script executel AGI cmd Dial to make another call.
Is there any reason not to have CDR record for the call that was initiated in
the AGI script?
Or I am just missing something basics .
Thanks,
--
You don't have
Cyrille DERORY wrote:
I'm using asteriskathome 2.0 beta 6 (asterisk 1.2) with SIP
softphones, 7905G cisco SCCP and analog phone( DTMF dialing). All is
working nice, however when I change DTMF for an analog pulse
dialing,my analog phone is not working.
I've found the following :
Hi
On Mon, Nov 28, 2005 at 04:35:34PM -0800, Geo wrote:
It should build wcfxo. Not trying anything special.
I just follow the procedure !
When I reboot I have:
ISDN subsystem Rev: 1.1.2.3/1.1.2.3/1.1.2.2/1.1.2.3/1.1.2.2/1.1.2.2 loaded
!!!
HiSax: Linux Driver for passive ISDN
Hi I'm trying to connect Avaya S8700 and Asterisk through H323 trunk
(using channel oh323).
I can make calls from S8700 H323 extension to Asterisk SIP phone using
G711a codec but when I try to make a call from SIP phone to S8700
extension I listen one ringing tone and the call is dropped.
Can
Hello,
I am trying to set up my dialplan in such a manner that calls to
numbers in the form 1-NPA-NXX- will only go through if the NPA
dialed is a geographical NPA in the continental United States.
I have collected a list of all NPAs that I want to allow, and have made
the following
On 11/28/05, Dustin Wenz [EMAIL PROTECTED] wrote:
According to the Asterisk wiki, adding the delete=yes option to a
voicemail definition should automatically delete messages after they
are emailed. This is the format that I'm using:
101 = ,First Last,[EMAIL PROTECTED],attach=yes|delete=yes
Hello,
While I was trying to get right CDR record from AGI script, I came across
cdr_manager.conf
I am trying to learn about cdr_manager.conf
What is the purpose of cdr_manager.conf?
How I can configure it?
I did google, really didn't have very good luck.
Would anybody please write couple of
On 11/28/05, Pablo Chacón [EMAIL PROTECTED] wrote:
Hi I'm trying to connect Avaya S8700 and Asterisk through H323 trunk
(using channel oh323).
I can make calls from S8700 H323 extension to Asterisk SIP phone using
G711a codec but when I try to make a call from SIP phone to S8700
extension I
I'm getting the following messages when a call is answered by a SIP device:
Nov 28 13:03:01 NOTICE[22824]: rtp.c:1146 ast_rtp_raw_write: RTP
Transmission error to 192.168.1.254:19262: Operation not permitted
For a Cisco 7940 line, I have the following sip.conf entry:
[desk2]
type=friend
Bob Goddard wrote:
It's not so bad... you do get access to firmware to all cisco devices
with that, so if you have more than one device it becomes worth it.
And it is also illegal.
Not true - that's the *point* of the more expensive contracts. They
cover you for each device that you own
Bob Goddard wrote:
I guess that should depend as to whether it is hardware or software only.
AFAIK all smartnet are software only... I've never heard of a hardware
contract.
(actually they're just an account on TAC which has access to certain
parts of the website - there's no physical part
On Monday 28 Nov 2005 20:41, Tony Hoyle wrote:
Bob Goddard wrote:
It's not so bad... you do get access to firmware to all cisco devices
with that, so if you have more than one device it becomes worth it.
And it is also illegal.
Not true - that's the *point* of the more expensive
On Monday 28 Nov 2005 20:42, Tony Hoyle wrote:
Bob Goddard wrote:
I guess that should depend as to whether it is hardware or software only.
AFAIK all smartnet are software only... I've never heard of a hardware
contract.
No, the vast majority of the smartnet contracts are hardware and
Hi BJ Weschke, thanks but unfortunately Ip address is the correct one.
Do you have S8700 with Asterisk working? using oh323 channel??
Maybe can help you my S8700 configuration...
My S8700 configuration is:
---
list
Hello All,
I'm using an Avaya 4620SW with Asterisk, the phone when hooked up to
the network, works for sometime, (I have not actually monitored the
time) maybe 20-30 minutes, after which the phone will still have a dial
tone, but can't dial out or recieve calls. I scanned thru the logs and
found
Sorry to reply to myself, but I need to add some information:
I have been informed (and now understand why) that the [] syntax does
not do what I had in mind here. Is there any syntax that will do it? If
not, I will just create a separate pattern for each NPA, which is
not a big deal, but I am
Hello Amir
On Sun, 2005-11-27 at 20:31 -0800, Amir Aziz wrote:
Dear List Members,
I am trying to setup a small asterisk box. My configure is pretty
basic for now. my zaptel.conf is as follows
[ ... ]
6. What other books/links can be helpful in learning this interesting
software.
I
I guess that should depend as to whether it is hardware or software only.
AFAIK all smartnet are software only... I've never heard of a hardware
contract.
Smartnet comes in serveral different flavors (eg, 24x7, 8x5) and
all of the flavors cover the hardware in addition to the software
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
hi,
very often, when the caller hangs up the phone, the isdn phone rings
without stopping. It seems, that asterisk does noch check, that the
caller has hang up.
I have this problem between ISDN-ISDN and ISDN-SIP. Is there a
solution for misdn?
cu
On 13:48, Mon 28 Nov 05, Sascha Deri wrote:
Additionally, the Retrieve voicemail butotn on the phones no longer
work. The MWI (Message Waiting Indicator) lights up, but when you press
the button you get Not Found sip:asterisk@ and busy signal.
I have been fighting with the same thing for
Greetings to all,
I am trying to get the line lights on a SNOM 320 to work using 'hint' in
extensions.conf. Unfortunately I have not been able to get it to work
properly.
Does anyone know for sure if the hint function works properly in 1.0.9?
If anyone has gotten this to work properly under
Hi.
Things are the same. I would be glad if you could help out.
Regards.
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Anyone know of a board, Digium, Sangoma or other, that supports QSIG?
Only hardware that I have seen that supports QSIG are Vegastream gateways.
Thanks in Advance!
--
Cory J Andrews
Partner / Purchasing
+++
VOIPSupply.com - Everything you need for VOIP
454 Sonwil Drive
Buffalo, NY
On Nov 28, 2005, at 3:00 PM, John Novack wrote:
Cyrille DERORY wrote:
I'm using asteriskathome 2.0 beta 6 (asterisk 1.2) with SIP
softphones, 7905G cisco SCCP and analog phone( DTMF dialing). All
is working nice, however when I change DTMF for an analog pulse
dialing,my analog phone is
From memory (at a previous installation) you will need a newer version of
Asterisk than 1.09 for the lights to work.
PaulH
- Original Message -
From: Joseph Rothstein [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, November 29, 2005 8:32 AM
Subject: [Asterisk-Users]
Bob Goddard wrote:
You misunderstand. Buying a smartnet contract for a phone does not
give you the right to download software for other hardware. One
smartnet contract equals only one device covered.
News to me... If cisco queried my TAC request of a router on an existing
contract bought
On 11/28/05, Pablo Chacón [EMAIL PROTECTED] wrote:
Hi BJ Weschke, thanks but unfortunately Ip address is the correct one.
Do you have S8700 with Asterisk working? using oh323 channel??
Maybe can help you my S8700 configuration...
My S8700 configuration is:
That appears to have done the trick...I guess I expected some sort of
warning at the console if I had inadvertently malformed the parameter
string. It works now though, so it's all good.
Thanks for the help!
- .Dustin Wenz
On Nov 28, 2005, at 2:15 PM, Gonzalo Servat wrote:
On
Joseph Rothstein wrote:
Greetings to all,
I am trying to get the line lights on a SNOM 320 to work using 'hint' in
extensions.conf. Unfortunately I have not been able to get it to work
properly.
Does anyone know for sure if the hint function works properly in 1.0.9?
If anyone has gotten this
Rich Adamson wrote:
If you read the legal stuff that must be acknowledged when downloading
software (under any cisco contract), you can only legally download the
software for the stuff that you have under contract. But, I've never
heard them enforce the web acknowledgement with any company to
Hi,
I am a newbie, and I am setting up a simple system to share a PSTN
line with another location.
In the process of setting this up I am also testing the various codecs.
I am only able to get comedian voicemail (ie dialing 1234) to record or
playback messages if I use the GSM codec? Is
Hello.
I`m using asterisk 1.0.9 and it`s working fine until I disconect the WAN
interface. Then asterisk doesn`t work fine, doesn`t make any Dial() and
I don`t know where is the problem. When I connect the WAN interface all
start working fine.
I`m also using NAT in the same server.
I don`t know
On 08:48, Tue 29 Nov 05, [EMAIL PROTECTED] wrote:
From memory (at a previous installation) you will need a newer version of
Asterisk than 1.09 for the lights to work.
on 1.0.9 the lights work.
In this way:
person is on the phone: light is on
Person is not on the phone: light is off
since 1.2
On 11/28/05, Kevin Hanson [EMAIL PROTECTED] wrote:
Joseph Rothstein wrote:
Greetings to all,
I am trying to get the line lights on a SNOM 320 to work using 'hint' in
extensions.conf. Unfortunately I have not been able to get it to work
properly.
Does anyone know for sure if the hint
Hi Jason,
There are a couple of boxes on the market these days that
have the following ports:
FXO/ISDN line out to PSTN
2 - FXS - analogue phone (or fax)
WAN port for DSL
As well as wfifi.
Fritz WLAN FON box for example.
Quick design:
Asterisk server at HQ. Each remote
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