Hi
As there been any progress regarding the use of spandsp 0.3 with
Asterisk 1.2.13?
Last month there was a thread about how spandsp 0.3 and rxfax from
http://www.soft-switch.org/downloads/snapshots/spandsp
made asterisk crash.
Is there any resources on how to get spandsp 0.3 work with
Ok thanks, do you think that it isn't possible to do that
automatically from asterisk?
On 12/12/06, Pavel Jezek [EMAIL PROTECTED] wrote:
I think, that adhoc conferencing isn't possible in this way, instead you
should use meetme, ie.:
skinny user calls to user A and transfer his to meetme number
hi, all,
I have realized a dundi cluster,,,
the details, please read...
http://jefferychen1977.spaces.live.com/blog/cns!9E49EEC4251C4476!494.entry
Thanks,...
On 12/13/06, David Thomas [EMAIL PROTECTED] wrote:
On 12/12/06, JR Richardson [EMAIL PROTECTED] wrote:
1.) When a registration
Matt Gibson wrote:
Hi Pavel,
I tried to implicitly set qualify=no for the sip user, but am still
seeing the registering icon for like 10 minutes on the screen of the
7970. It is actually registering, just the phone doesn't think it is.
The phones always stay with a little red X on them
CDP has nothing to do with inline power, it is L2 proprietary protocol
for negotiation of voice vlan between phone and switch,
so you don't need to set what vlan number phone should use for voice and
what is for connected pc data.
if you disable cdp on switch, phone will still working, except
maybe some asterisk guru have idea for some smart script, how to do this
;-)
I found some RFC for better sip conferencing, but currently probably not
implemented in asterisk :'(
High-Level Requirements for Tightly Coupled SIP Conferencing
ftp://ftp.rfc-editor.org/in-notes/rfc4245.txt
Hi Mike,
Do you have a full SIP trace?
Cheers
Dave
Dave,
here is the trace, the BYE message at the end:
-- SIP read from 192.168.6.222:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.6.222:5060;branch=z9hG4bK-a993e341
From: 201 sip:[EMAIL
Hi
Is NAT set to NO?
It needs to be set to NO in 8.0.3 or it just sits there at registering as
you say
Thanks
- Original Message -
From: Pavel Jezek [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday,
[Description]
Busy([timeout]): This application will indicate the busy condition to
the calling channel. If the optional timeout is specified, the calling
channel
will be hung up after the specified number of seconds. Otherwise, this
application will wait until the calling channel hangs up.
Hi.
After successfully running ./configure I run make. When running make I get
the
following error..
[CC] ast_expr2f.c - ast_expr2f.o
[CC] ast_expr2.c - ast_expr2.o
[CC] strcompat.c - strcompat.o
[LD] aelparse.o aelbison.o pbx_ael.o ael_main.o ast_expr2f.o ast_expr2.o
strcompat.o -
Hi.
After successfully running ./configure I run make. When running make I get
the
following error..
[CC] ast_expr2f.c - ast_expr2f.o
[CC] ast_expr2.c - ast_expr2.o
[CC] strcompat.c - strcompat.o
[LD] aelparse.o aelbison.o pbx_ael.o ael_main.o ast_expr2f.o ast_expr2.o
strcompat.o -
there is any way to configure a 7970 without using the display, I have
my LCD broken so I cannot see what I'm doing :) but the phone works
fine.
2006/12/13, Paul A Brown [EMAIL PROTECTED]:
Hi
Is NAT set to NO?
It needs to be set to NO in 8.0.3 or it just sits there at registering as
you say
Hello,
I have installed asterisk version 1.2.12 and latest zaptel modules. but i
can see some IRQ conflicts on the server. iam uisng two TDM04B cards.
according to my previous knowledge on asterisk verison 1.07 asterisk has
given lot of erros when starting if you have assigned the same IRQ number
Christophorus Laube schrieb:
[Description]
Busy([timeout]): This application will indicate the busy condition to
the calling channel. If the optional timeout is specified, the calling
channel
will be hung up after the specified number of seconds. Otherwise, this
application will wait until the
Hi Pavel,
Thanks for the config!
I modified mine so it was more minimal like yours, and it registers
just fine now. So much nicer without those big red X's!
MG
On 13/12/06, Pavel Jezek [EMAIL PROTECTED] wrote:
Matt Gibson wrote:
Hi Pavel,
I tried to implicitly set qualify=no for the
Am Mittwoch, den 13.12.2006, 11:47 +0100 schrieb Fabian Foerster:
Is there any output on the CLI that proves the BUSY command is run at
all? Because I don't really know if
exten = _X.-BUSY,4,Busy(1)
is gonna work. I would say something like:
exten =
Hi,
In what Asterisk file can I edit for me to temporarily unload such modules. But
later I woudl still be able to load them.
Thanks
Angel
Yahoo! Music Unlimited
Access over 1 million songs.
Hi,
You may want to visit www.procurve.com and look for thier training section
there are lots of training materials that can be downloaded. Prices are also
posted in this website.
Actually, all networking manufacturers has thier training docs posted in their
websites.
www.3com.com
Hi,
Actually on my setup all outgoing calls are going trhu a SIP unique account
A have a second SIP account with another operator and I would like my setup
to use alternatively each of the two accoutns
Call 1= Dial SIP/phone1
Call 2= Dial SIP/phone2
Call 3= Dial SIP/phone1
...
If you have an
Hi,
Actually on my setup all outgoing calls are going trhu a SIP unique account
A have a second SIP account with another operator and I would like my setup
to use alternatively each of the two accoutns
Call 1= Dial SIP/phone1
Call 2= Dial SIP/phone2
Call 3= Dial SIP/phone1
...
If you have an
/etc/asterisk/modules.conf
On 12/13/06, Angel Heart [EMAIL PROTECTED] wrote:
Hi,
In what Asterisk file can I edit for me to temporarily unload such
modules. But later I woudl still be able to load them.
Thanks
Angel
--
Cheap Talk? Check
Hi
i use now iaxmodem for receive fax and that's work very good with
Hylafax ;=)
Do you know if we can sent fax using iaxmodem and Hylafax ?
when i test:
déc 13 13:47:21.12: [13725]: SESSION BEGIN 00014 330426690268
déc 13 13:47:21.12: [13725]: HylaFAX (tm) Version 4.3.0
déc 13
On Wed, Dec 13, 2006 at 02:03:09AM -0800, Angel Heart wrote:
Hi,
In what Asterisk file can I edit for me to temporarily unload such
modules. But later I woudl still be able to load them.
Works fine as long as the module is not in use.
asterisk -rx 'unload app_test.so'
Later on:
Sorry, sorry !!! I was mixed with another config when I wrote my first
email !!
In fact, User A is registered on Asterisk and user B has a public phone
number (no link with Asterisk).
Our test is : User A calls asterisk server via SIP. As User A context
has a DIAL('user B phone number'),
On 12/13/06, Savoy, Kevin - Williston, ND [EMAIL PROTECTED] wrote:
I am trying to set up a Conference room where users are put on hold
until the host arrives. I have figured out that the A option activates
marked mode and the w option is used to activate the waiting until the
marked user
Hi Guys,
I'm using Asterisk with Hylafax to send and receive faxes, currently only
receinving with success.
When sending i get this:
Dec 13 11:28:07.51: [ 9242]: SESSION BEGIN 00157 03510212079
Dec 13 11:28:07.51: [ 9242]: HylaFAX (tm) Version 4.3.1
Dec 13 11:28:07.51: [ 9242]: SEND FAX:
Am Dienstag, den 12.12.2006, 13:08 -0600 schrieb Savoy, Kevin -
Williston, ND:
I am trying to set up a Conference room where users are put on hold
until the host arrives. I have figured out that the A option activates
marked mode and the w option is used to activate the waiting until the
Talk to your carrier. Most likely you won't be able to hide it. You
might be able to set it.
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michel
Sent: Wednesday, December 13, 2006 7:56 AM
To: asterisk-users@lists.digium.com
Subject:
Michel wrote:
Sorry, sorry !!! I was mixed with another config when I wrote my
first email !!
In fact, User A is registered on Asterisk and user B has a public
phone number (no link with Asterisk).
Our test is : User A calls asterisk server via SIP. As User A context
has a DIAL('user B
I wonder if anyone can help me with this. I have 4 sites running
asterisk and calls coming into any of these sites are received locally
and forwarded to a central operator. E.g. Call comes in on site A and
is forwarded to the operator on site B. 99/100 the operator will send
the call back to
On Wed, 2006-12-13 at 15:53 +1100, Paul Hales wrote:
What does zttool show?
And after you 'modprobe wctdm' what does your dmesg
read? /var/log/messages?
You should see something about a card being recognised
PaulH
After I modprobe wctdm, nothing new shows up in /var/log/messages
I forgot to mention that the feature in question is called CLIR, or
Calling Line Identification Restriction. With that, you can always hide
the presentation of your caller ID or do that on a per-call basis. You
might want to ask your telco about that.
Hello peoples,
I need to do a test of urgent stress. It know as much as connections
simultaneous my equipment is going to do passing codec g729 and g723.
Someone knows say me as obtain does him?
Andre Luiz Martins
mailto:[EMAIL PROTECTED]
___
I'll give this a try but seems silly to require 2 different extensions
for one conference room. Thanks for the input.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anselm
Martin Hoffmeister
Sent: Wednesday, December 13, 2006 7:16 AM
To: Asterisk Users
I did try this and it doesn't work. When logging in with the admin
password it still waits for the marked user.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of RR
Sent: Wednesday, December 13, 2006 7:03 AM
To: Asterisk Users Mailing List - Non-Commercial
IF I wanted to do the whole sophisticated telephony VoIP stuff
asterisk, what hardware would I need? I have a feeling that my fax
modem is probably not going to work out. My wife and I have an income
of $650 a month. After the first-of-the-month bills are payed, we're
lucky if we have $100 left
I was able to get it to work with 2 extensions. One for the host and
one for the participants Not the best way to set it up but it works.
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Savoy,
Kevin - Williston, ND
Sent: Wednesday, December 13,
cheapy PC (throw away PII is fine) and if you want to use the PSTN, a
X100P FXO card. These can be had on ebay for 11 bucks, but I understand
that even that pushes the bank some days. You don't need the card, you
only need it if you want to receive or place calls on the PSTN. You can
use
On Wed, 2006-12-13 at 08:29 -0600, jason wrote:
cheapy PC (throw away PII is fine) and if you want to use the PSTN, a
X100P FXO card. These can be had on ebay for 11 bucks, but I understand
that even that pushes the bank some days. You don't need the card, you
only need it if you want to
Just emailing the list to see if anyone out there has used Vitelity? If so
what has been your experience with service, support etc?
Thanks
Curt
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or
Don't forget that IF you have NO card, you need to roll ZTDUMMY into the
compile. With no card though, you will not be able to read the incoming CLID
Also, IF you ever want to progress beyond the X100 card, The Digium
cards ( beyond your present budget ( are really intolerant of older PCI
Hi guys. This is my 1st post here (after much reading). I have a test
asterisk system setup using X-Lite Soft Phones, and the issue I am
running into is that caller id shows up as asterisk on all incoming
calls and on all local to local calls (internal). I have showcallerid,
etc.
Anyone have any experience with the Audiocodes MediaPack MP-118? We are
looking at options for a location that wishes to maintain 6 - 8 existing
analog phones in addition to a number of new Polycom phones.
Thanks,
Mike Clark
___
--Bandwidth and
nope, just a regular old phone cord. with that card and a PC, you can
receive calls, dial out, terminate SIP, IAX, create an answering
machine, run voicemail, talk to jabber servers, all kinds of fun stuff!
Asterisk is almost as good as Legos and a lot easier on bare feet at 2am!
Michael
You can start off using Soft Phones on your PC (they are free) at 1st once
your happy that you want to play voip then you can get either a VOIP hard
phone or a VOIP to analog adaptor (Analog Telephone Adaptor), the latter
provides you with an FXS port that you can plug a normal phone into or
Greetings, all.
I would like to record calls that are entered into queues and I'm not
quite sure how to do it. Here's how I'm currently set up:
- Call comes in and is placed into Queue #1 (which rings all phones for
15 sec).
- If call drops out of this queue, it is placed into Queue #2
We're using vitelity, not in large scale call center type numbers, but
any long distance numbers we dial go out their system. They've been
working great, but if you expect support for an asterisk system, don't
bother calling them. The furthest they'll go is telling you that there
are configs on
The card will let you interface with a regular telephone line instead
of VoIP. If you want to use a regular phone instead of the computer
softphones, look into the Grandstream handytone devices - they'll
make it so your regular telephones can talk to Asterisk. You can
make the system
Has anyone ever gotten the Polycom Status feature, accessible via the 'MyStat'
soft-key to work? When you change the status in this way, the phone does not
send any communication to Asterisk, and it seems to have no effect in incoming
calls. So... what's it for?
Doug
Hi Gordon
I too have this problem with one of my two BT lines, very very annoying. I
was using a TDM400P and am now using a Sangoma A200 (lightning got the TDM),
I think the sangoma is very slightly better (less echo) but I might just be
kidding myself. Another think I've found it that using a
Speaking of the X100P, I am going to setup an asterisk server next
week for a friend's business to replace his aging system. He
currently has two voice lines and another line for the fax machine.
I was looking at the Sangoma A20200D but that's pretty expensive...
We're going to use
I have a development box connected to them and place calls on it from time
to time and let family members use it. I have never had any problems, but my
usage is rather light and outages might not be noticed with the low volume
of calling.
On 12/13/06, Curt Shaffer [EMAIL PROTECTED] wrote:
Just
You may want to have a look here: http://astrecipes.net/index.php?n=42
Best regards
l.
On Wed, 13 Dec 2006 16:15:17 +0100, Jay Moore [EMAIL PROTECTED]
wrote:
Greetings, all.
I would like to record calls that are entered into queues and I'm not
quite sure how to do it. Here's how I'm
In queues.conf you must have the following under the queues you want to record.
monitor-format=wav49 ; you may also use wav or gsm formats
monitor-join=yes; if you have the latest sox installed,
thiswill join the in and out
I've been trying to find where to download the Web Vmail application and
instructions on how to install it for Asterisk BE. Any ideas?
Thanks
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
-Original Message-
On 12/13/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Has anyone ever gotten the Polycom Status feature, accessible via the
'MyStat' soft-key to work? When you change the status in this way, the phone
does not send any communication to Asterisk, and it seems to have no effect
in incoming
Matt Gibson wrote:
Hi Pavel,
Thanks for the config!
I modified mine so it was more minimal like yours, and it registers
just fine now. So much nicer without those big red X's!
MG
This modified config works sweet!! Any tricks to get the MWI working?
Mark
I think the only gotcha on them is the strange convergence of EXGN and
Sixtel that resulted in Vitelity. But hey, maybe they combined their
strengths. That said, my sixtel experience was lousy, my EXGN
experience ok, and so far, I don't have any real complaints with
Vitelity.
We have been testing them for about a month on outbound only. All I have to
say is good luck getting setup. It took us several months just to get a test
account and now that we want to actually get service we can't get anyone
over there to return our e-mails or calls. They are great for calls but
Am Mittwoch, den 13.12.2006, 15:03 + schrieb
[EMAIL PROTECTED]:
Hi guys. This is my 1st post here (after much reading). I have a test
asterisk system setup using X-Lite Soft Phones, and the issue I am
running into is that caller id shows up as asterisk on all incoming
calls and on
On Wed, 13 Dec 2006, Wireless wrote:
Hi Gordon
I too have this problem with one of my two BT lines, very very annoying. I
was using a TDM400P and am now using a Sangoma A200 (lightning got the TDM),
I think the sangoma is very slightly better (less echo) but I might just be
kidding myself.
For my home phone system I have an old P-II, which is working perfectly fine
for last more than a year now. I had a P-III before that, but one day it
died. This P-II is still working and we have no problems with our phone
system. I even had conference calls on it with 6 simultaneous users. For
Hello.
I'm trying to setup MFC/R2 signaling, but chan_unicall fails at loading
with this message:
---
Unable to load module chan_unicall.so
Loaded /usr/lib/asterisk/modules/chan_unicall.so = (Unified call
processing (UniCall))
== Parsing '/etc/asterisk/unicall.conf': Found
061213-075938
Hoping someone out there has run into this or has some ideas for us.
We currently have asterisk set up with Realtime (using mysql) for its
extensions,sip and voicemail files.
The problem we are trying to solve, is one of a failover mechanism. What
if our mysql server went down.
Can Realtime be
It still has to go through the upstream pbx/proxy. Each phone doesn't know the
location, ie ip address, of the other phones. When the state changes, it should
send an updated SIP subscription to Asterisk.
-Original Message-
From: LST [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December
Does anyone have the pickup application working? I'm attempting to get
it so that a particular extension programmed into a phone can be picked
up by another phone with that extension programmed with a speed dial
with a 'p' in front... basically, if you dial p100 and extension 100 is
ringing,
Fyi... My apologies if this is a dupe.
-Original Message-
From: Cisco Technical Support
[mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 13, 2006 8:52 AM
To: Tim Connolly
Subject: New Software available on Cisco.com
New software images are available on Cisco.com for the product
On 12/13/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Has anyone ever gotten the Polycom Status feature, accessible via the
'MyStat' soft-key to work? When you change the status in this way, the phone
does not send any communication to Asterisk, and it seems to have no effect
in incoming
I have correct Caller-ID information coming in on the 'Remote-Party-ID' header.
The From value is being sent in as Unknown.
How could I replace the From value , or CALLERID(all) with the correct
values that are in Remote-Party-ID? Or is there a way to tell asterisk
to read that header?
Hi,
I have an asterisk with a DIVA Server V-BRI-2 card, connected to a Siemens
PABX. From a SIP phone, I can call other internal SIP phones, external
numbers (to PSTN), but I can't call internal phones connected to the
internal phone network.
When I call 107, which is an internal phone,
Anyone seen this...? Is it a known issue?
I'd file a bug, but we're on 1.2.9.13, and every time I file a bug and it isn't
against the latest code I get given crap for it. Given that most of the time
you don't know HOW to reproduce a problem on the latest code anyway, not
accepting bugs from
Anyone able to point me the right direction for the following would be
helpful.
I have a client that needs to keep detailed time for how long their Customer
Service Reps. Spend on different subject with the customers.
i.e.
All CSR's are trained to take all types of calls.
For regulatory
I'm looking to use * for a HQ/branch office topology with fairly few calls
over the WAN. The questions I have all pertain to the following
architectural pic: http://www.45891.com/misc/arch.jpg
I'm looking at a distributed architecture so users are somewhat functional
when the link to HQ is
On Wed, Dec 13, 2006 at 09:12:13AM -0500, Alejandro Rios Peña wrote:
Hello.
I'm trying to setup MFC/R2 signaling, but chan_unicall fails at loading
with this message:
---
Unable to load module chan_unicall.so
Loaded /usr/lib/asterisk/modules/chan_unicall.so = (Unified call
processing
I saw on a mailing list for digium that back in March, they were looking
to get SRV working properly.
Was this ever repaired? If so, is it just a matter of adding 2 entries
to tinydns data file, and then point the res_mysql.conf file to point to
the new hostname (astmysql.yournet.com)?
Trying
Hello,
does anybody know if it is possible to save the IP address of an IAX
client logging into asterisk into the DB for future reference?
I.e. one could distinguish between cases, where the client was last seen
on the local net or on the road... even when it is not currently online.
Hello
I would like to define a trunk from my Asterisk to a VoIP provider, but
I want to make it secure, because its through the Internet.
I want to be sure no one makes calls as being me, and that my calls
aren't intercepted.
Is it possible to define encrypted trunks? And should I define the
On 12/13/06, Rob Schall [EMAIL PROTECTED] wrote:
Hoping someone out there has run into this or has some ideas for us.
We currently have asterisk set up with Realtime (using mysql) for its
extensions,sip and voicemail files.
The problem we are trying to solve, is one of a failover mechanism.
At 04:27 AM 12/13/2006, you wrote:
A have a second SIP account with another operator and I would like my setup
to use alternatively each of the two accoutns
Call 1= Dial SIP/phone1
Call 2= Dial SIP/phone2
Call 3= Dial SIP/phone1
...
If you have an sample please let me know
Something like this
On 12/13/06, Rob Schall [EMAIL PROTECTED] wrote:
I saw on a mailing list for digium that back in March, they were looking
to get SRV working properly.
Was this ever repaired? If so, is it just a matter of adding 2 entries
to tinydns data file, and then point the res_mysql.conf file to point to
iftop
On 12/12/06, Mochamad Susantok [EMAIL PROTECTED] wrote:
Dear all,
Are there anyone have ben to use some tool or method to measure latency
and packet loss for VoIP packet ?
-
This email was sent using Student EEPIS-Webmail.
Anselm Martin Hoffmeister wrote:
Am Mittwoch, den 13.12.2006, 15:03 + schrieb
[EMAIL PROTECTED]:
Hi guys. This is my 1st post here (after much reading). I have a test
asterisk system setup using X-Lite Soft Phones, and the issue I am
running into is that caller id shows up as asterisk
I've added the ability for a user to record a custom message associated
with a special IVR menu for occasions when business will be closed for
some non-standard amount of time (Maybe 4 days at Christmas...) They
just dial 800, record the message then hang up and dial 801 to enable
it.
http://www.voip-info.org/wiki/view/IAX+encryption
Joao Pereira wrote:
Hello
I would like to define a trunk from my Asterisk to a VoIP provider,
but I want to make it secure, because its through the Internet.
I want to be sure no one makes calls as being me, and that my calls
aren't
John French wrote:
I've added the ability for a user to record a custom message associated
with a special IVR menu for occasions when business will be closed for
some non-standard amount of time (Maybe 4 days at Christmas...) They
just dial 800, record the message then hang up and dial 801
Hi Lan -
I am trying to upgrade my testing asterisk realtime 1.2.13 with MySQL 5.0
and unixODBC to the beta asterisk 1.4.
I run the make and make install for the asterisk-addon just fine, It created
the modules res_config_mysql.so and cdr_addon_mysql.so without any problem
or error. However,
It is possible to announce the parking position through a paging to a group
of extensions?
I would like that when someone parks a call, some phones will announce with
the speaker the position.
Something like:
exten = s,1,ParkAndAnnounce(call-parked-at:PARKED|30|PAGE(LOCAL/[EMAIL
Hi everybody...
I have a similar problem... I don't get the ID of the person that i
called on my phone... Does anyone know something about this problem?
greets,
Sven
2006/12/13, Bruce Ferrell [EMAIL PROTECTED]:
Anselm Martin Hoffmeister wrote:
Am Mittwoch, den 13.12.2006, 15:03 + schrieb
When the IP601 is sitting unused, it uses the first 2 of the 4 soft
buttons under the screen. The third one is empty, which is good because
it is used for Exit.
I would like to be able to use that 4th button for group pickup (*8#)
and have it read Pickup. Is this possible? If so, how?
Is anyone else having trouble getting a Polycom IP4000 (running SIP
1.6.7 and BootROM 3.1.3) to download its configuration files from a
vsftpd 2.0.1 server? We have 100+ IP501s that manage this without
problems, but the IP4000 keeps timing out.
We have opened a case with Polycom, but they are
In article [EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote:
Anyone seen this...? Is it a known issue?
I'd file a bug, but we're on 1.2.9.13, and every time I file a bug and it
isn't against the
latest code I get given crap for it. Given that most of the time you don't
know HOW
On 12/13/06, Ed Nuñez [EMAIL PROTECTED] wrote:
I've been trying to find where to download the Web Vmail application and
instructions on how to install it for Asterisk BE. Any ideas?
Is this any different than the vmail.cgi that comes with the open version?
Otherwise, you will just need to
Carlos Rojas wrote:
iftop
On 12/12/06, *Mochamad Susantok* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Dear all,
Are there anyone have ben to use some tool or method to measure
latency
and packet loss for VoIP packet ?
Commercial or Open Source?
For Open Source,
Do you have the latest firmware files from polycom and sample
configurations? Can you get the phone to accept those? Any reason why you
are using FTP? Http has worked without a hitch. What does your logs say?
On 12/13/06, Anthony Rodgers [EMAIL PROTECTED] wrote:
Is anyone else having trouble
How about put it in the dial plan? So anytime you try to make an
outbound call it would play a reminder saying that the alternate
greeting is enabled. You could just use a DB variable.
On 12/13/06, Mailinglisten [EMAIL PROTECTED] wrote:
John French wrote:
I've added the ability for a user
Hi Warren -
When the IP601 is sitting unused, it uses the first 2 of the 4 soft
buttons under the screen. The third one is empty, which is good because
it is used for Exit.
I would like to be able to use that 4th button for group pickup (*8#)
and have it read Pickup. Is this possible? If
On Wed, 13 Dec 2006, Gregory Duchatelet wrote:
Hi,
I have an asterisk with a DIVA Server V-BRI-2 card, connected to a Siemens
PABX. From a SIP phone, I can call other internal SIP phones, external
numbers (to PSTN), but I can't call internal phones connected to the
internal phone network.
Has anyone done this, or have a thought on how to do it.
I forsee it working like this...
Dial in to a main greeting, dial an extension using a modem string like
782-,,,##409*. The extension would some kind of modem emulator. I
know this compromises security. I was hoping to use an
Hi All -
I haven't worked much with ZAP handsets before, but I've got a client
who is insistent on using a particular phone. My questions:
1. With multiline analog phones, if I've got multiple phones, each
connected to a different FXS interface, is there a way to make the
line status lights on
Hi Jordan -
Has anyone done this, or have a thought on how to do it.
I forsee it working like this...
Dial in to a main greeting, dial an extension using a modem string like
782-,,,##409*. The extension would some kind of modem emulator. I know
this compromises security. I was hoping to
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