Trying to get some straight info from the VOIP providers is difficult.
Say there's a small Asterisk switch and it's registered with Broadvoice
or LiveVOIP or someone. There are a couple of people using the switch,
one is on an outgoing call with the VOIP provider. What happens when
someone
I'm trying to set up extensions and have them forward to my cell phone
or other phones I have and include them in call groups.
I tried *72480204 and *7298480204, I get the recording that
unconditional forwarding is set to that number..
but when I call that extension I just get silence
On Fri, 25 Mar 2005, Jessie Mabanglo wrote:
You can look for Plantronics.. we're been using here in our call center for
2 years...they have a variety of models...
Our call-center were allowed to test quite a few borrowed headsets.
Sennheiser telecom headsets generally came up on top when
I suggest you strip naked, do a war dance, and sacrifice chickens to
the digium gods, and then i'm sure verything will work fine. If that
doesn't work, do the same thing while standing on your head.
Or, you could post some details of your installation so we have some
faint idea of what might
If we need a dose of Smart Ass, it's always good to know it's available
here on this list. The person is new and he is asking a question. You could
have emailed him direct, asked for more detail and helped him. Rather than
be kind you posted dribble. Daily I speak with people like Trevor. If
On Fri, 25 Mar 2005 17:33:17 +1000, Greg
[EMAIL PROTECTED] wrote:
I am trying to convert my 7905G to be SIP based and seem to be running
into a few hassles. Below are all the config files and logs from the
server. I have tried to follow the pdf's from cisco and some posts from
other mailing
Jeez, people...learn to take a joke.
I offered to help the man. I didn't make cracks about googling or
anything like that. I said explicitly that if he posted details we'd
try to help. I didn't insult him, call him a worthness noob, or
otherwise offend him. IT WAS JUST A JOKE. Is this different
Our main asterisk box peers with that of a customer. We are trying to assign
DID's to their extensions but get this error. What are we doing wrong?
Client side
Mar 25 18:49:47 NOTICE[1369]: chan_iax2.c:6545 socket_read: Rejected connect
attempt from 203.xxx.xxx.16, who was trying to reach 's@'
Agreed - So lets help the new Guys!
Brandon Patterson
LiveVoip LLC
Jeez, people...learn to take a joke.
I offered to help the man. I didn't make cracks about googling or
anything like that. I said explicitly that if he posted details we'd
try to help. I didn't insult him, call him a worthness
On Thu, 24 Mar 2005 21:26:27 -0800, Max Clark [EMAIL PROTECTED] wrote:
Hi all,
Good evening
I have a working (it was a pain) set of Cisco 7960 phones. In order to
dial I have to either pick up the handset or select the line and then
dial the extension or outside line. How do I configure the
What happens if a SIP call is routed through more
than one * server?
If canreinvite=yes for all the peers involved, and t or T is not used in
the Dial command, then the audio would get routed directly between the
endpoints.
Also, when setting up an inter asterisk exchange, is all the
data
http://www.loligo.com/asterisk/Cisco/79xx/current/asterisk-tux.bmp
On 25/03/2005, at 7:01 PM, Shaun Ewing wrote:
A while ago I found a cool asterisk/penguin logo to use on the phone,
can anyone point me to a place I can download this again?
Wouldn't have a clue, but would also like to know :)
Yair Hakak wrote:
Jeez, people...learn to take a joke.
Use your smiley button, then. Lots of people have pretty short fuses on
both sides of this issue, and it's well to avoid ambiguity whenever
possible.
Your post was ambiguous in that respect, where for three extra
keystrokes you could
Thanks!
That looks like what I need. I just want caller ID to appear on every
handset. I have wireless phones too and (fortunately) there is no Outlook
on those phones :) but I would like CIDName.
Cheers!
Remco
On Thu, 24 Mar 2005, Jay Milk wrote:
Export Outlook to CSV, import name and numbers
On Thu, Mar 24, 2005 at 05:36:35PM +0200, Mark Elkins wrote:
I am still curious. Which Driver do you use for the HFC card?
I manage my own Debian package repository for Debian stable (woody)
backports of asterisk and related stuff (based somewhat on
backports.org).
I currently use:
Version:
What about imaging?
We use acronis true image 8.0.
You can create an image of your asterisk box within 20 minutes (120 GB HD !)
and deploy it to another server in the same time. Even if changing your
hardware from VIA to SIS and back to INTEL wasn't a problem for us.
Btw we use Fedora Core 2 for
Hi All,
Thanks for the wonderful advice, and comments, and anything I might of
missed, and no offence taken.
Yes I am new to this program, and Linux too, so this is a big learning
curve.
I installed the software Asterisk which I believed it did straight from
the cd, rebooted the computers, and
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi folks,
I've some questions about asterisk, and in general about voip, please
help me :)
1. I've SIP accounts on external servers, and I would that my local
server will connect with those and redirect all calls from those to an
internal SIP
Richard,
Yep, got that config'd in there:
1001 = 1001,Andy Stewart,[EMAIL PROTECTED]
1002 = 1002,Lorri Barnett,[EMAIL PROTECTED]
1003 = 1003,Andy Stewart - Home,[EMAIL PROTECTED]
1004 = 1004,Andy Stewart - HardPhone,[EMAIL PROTECTED]
1005 = 1005,Lorri Barnett - HardPhone,[EMAIL PROTECTED]
Jim Singh wrote:
In our setup, outbound call volume frequently exceeds
the line capacity of the DSL line. We do not want to
move to another codec to better utilize the line, but
instead wish to automatically divert overflow to the
Long Distance T1 when the DSL is full. Ideally the
system would
Mark Charlton wrote:
Hi all
I want to send an SMS message whenever I get a voicemail left on my [EMAIL
PROTECTED]
0.6 box. I don't have any pstn attached the the box, and I am running FWD,
voipuser, and alg as providers for various routes and redundancy. I can
find a number of providers for
Hi,
a few pointers:
1. the wiki is your friend:
http://www.voip-info.org/wiki-Asterisk
lots of good stuff and good documents for getting started. If i
were you i might reinstall asterisk from CVS just to make sure you
have the latest version, and because this way you can learn about
On Thu, Mar 24, 2005 at 01:30:25PM -0500, Joel Duffield wrote:
I tried to share my spool directory so I could get monitored calls, and now
this error comes up when I try to leave a message in any of my voicemail
boxes.
Mar 24 12:48:35 WARNING[344081]: file.c:906 ast_writefile: Unable to open
Hi list,
This wll be my first post, so I want to thank all the developers for the
great product they have created.
Now, the question,
I have installed asterisk 1.05 on debian sarge (binary package)
with an I4l modem and 4 x-lite softphone and 2 SIP hardphones (Yuxin 100)
This all works fine,
Hi,
Does IAX supports silence suppression? If yes, is there any way to
detect that the other party has turned on silence suppression and there
is no packet loss? Is (Halt|Reasume) audio/video transmission control
messages used for this reason?
Regards,
Marcin Okraszewski
would like to test this e-mail list.
anyway, have anybody here install and run [EMAIL PROTECTED] how was it?
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To UNSUBSCRIBE or update
Short answer is no. You should always turn it off on any client you
have.
Longer answer is that is is being worked on and should be available any
day now (although that has been the case for some months).
Also someone is working on porting it to SIP as well as IAX2.
No idea if the new work will
Hello List,
I'm trying to setup MGCP channel with a Centile Media Hub box. My
Centile box has 4 ports and I got no dial tone. Can somebody help with
this isuue?
This is my mgcp.conf and extensions.conf
Thanks
Daniel.
; MGCP Configuration for Asterisk
;
[general]
port = 2427
bindaddr =
+++ Dan [20/03/05 09:17 +0200]:
Hi James,
- Original Message -
From: James Coberly [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, March 19, 2005 11:41 PM
Subject: Re: [Asterisk-Users] Re: Optional URL
I have some numbers, which should be treated equally. To avoid double
coding, I would like that this extension could be re-written.
E.g., some users are used to dial 002 ~ 009 as international prefix,
while I have choosen to use the USA way (011).
It would be nice if the user can dial
Hi,
Thanks for the pointers, I will do a reinstall and see how I go.
I am presuming that this program will work with just the x-lite and no
others phone related hardware.
I am just wanting to get it working from pc to pc to start with, then will
attach the next step of going to a live phone
Welcome,
Yes I have used it. It's great to get started. Give it a try.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bagan Jermal
Sent: Friday, March 25, 2005 6:34 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Hello Everyone
would
Version 1.0.5.23 is now available from http://gs-firmware.gratissip.dk/
Or directly from Grandstream at
http://www.grandstream.com/BETATEST/Release-b21p1.0.5.23.zip
Release notes doc here
http://www.grandstream.com/BETATEST/Release_Note_1.0.5.23.doc
while on the matter I just want to extend a
Stewart Nelson wrote:
How should I proceed? IMO, this provider offers an excellent combination
of price, reliability, quality, and support, and I believe that many in
Asterisk community would want to use them. AFAICT, their SIP/SDP does
not actually violate any RFCs.
The next step would to be
Look at the options for the dial command on the wiki, you have to use t
or T or calls are not eligible to be transferred.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of asterisk
Sent: Friday, March 25, 2005 3:54 AM
To: Asterisk Users Mailing List
Looks like you do not have the context set correctly in your iax.conf on
both sides. Make sure that it exists and it is going to do what you want
it to do.
On Fri, 25 Mar 2005 18:57:06 +1000
AS [EMAIL PROTECTED] wrote:
Our main asterisk box peers with that of a customer. We are trying to
Yes Andy - that was my mistake. I have my system hacked up to do some
other things.
It should be:
1234 = 1234,Bob Jones,[EMAIL PROTECTED]
do your mail logs have any errors at all in them in regards to mail
bouncing or anything like that..?
Do you have your servermail settings configured in
Hello.
Sorry for my bad english. I'm a french guy.
I have the same problems with siemens dect phones S100
The caller id don't work on tdm...
In France, the CID is differant than other country.
Then standard ring cadence is: 1500 3000 1500 3 and so...
The etsi standard (used in France) say:
Hello
I want to to know if the motherboards VIA are fully supporte by asterisk.
And also, some of those motherboars say that with 1 pci slot , using a
special riser card you can connect 2 pci cards. Will that work to have 2 pci
cards (FXS or FXO ) on asterisk?
thank you
Fabian
Hi Noah -
I've managed to get my asterisk server up and running with a single
POTS
line and a polycom IP500.
It will happily answer the phone line, tranfer calls, voicemail, etc.
The problem comes when I pick up the polycom phone and want to place an
outside call.
If I dial 913237773456 it just
I do, like I do with my IAX2 softphone. It's just that I haven't
tookthe time to make a webpage that explains what it does and provide
alink to download it.I already send it to peoples on this list that
asked for it.Anybody want it, just email (privately, since this list is
already
Sorry about the previous post. Is this still available? The main
thing is I need a management tool I can use in commercial sales.
Regards,
Chris
[EMAIL PROTECTED]
Original Message
I do, like I do with my IAX2 softphone. It's just that I
I gave up on tape as being a nightmare to maintain, I now back all my
servers and workstaions using backuppc. One linux server with a 5 device
RAID can easily backup 100 workstatons and several servers beacuase of the
pooling system used. For a smaller situation I would use 2 disks in RAID1
On 25 Mar 2005, at 14:35, Chris Mason wrote:
I gave up on tape as being a nightmare to maintain, I now back all my
servers and workstaions using backuppc. One linux server with a 5
device
RAID can easily backup 100 workstatons and several servers beacuase of
the
pooling system used. For a
Hi all, I have an Athlon 64 server with Fedora Core 2 x86_64. When I try
to make asterisk-addons-1.0.7 (and olders) it can't and it say me
[EMAIL PROTECTED] asterisk-addons-1.0.7]# make install
./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c`
make -C format_mp3 all
Message: 16
Date: Fri, 25 Mar 2005 01:06:21 -0700
From: JD [EMAIL PROTECTED]
Subject: [Asterisk-Users]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
I'm
Hello
I have setup [EMAIL PROTECTED] and can login
to the system via the asterisk box. But if I try same username and
password to login using the Asterisk Management Portal I try the same username
and password and cannot login. says authorization failure. I have
tried from a Windows 2000
On Friday 25 March 2005 10:09, Jeff R Glassman wrote:
Message: 16
Date: Fri, 25 Mar 2005 01:06:21 -0700
From: JD [EMAIL PROTECTED]
Subject: [Asterisk-Users]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
David Josephson wrote:
Trying to get some straight info from the VOIP providers is difficult.
Say there's a small Asterisk switch and it's registered with
Broadvoice or LiveVOIP or someone. There are a couple of people using
the switch, one is on an outgoing call with the VOIP provider. What
Title: Message
i
believe the login for AMP is username:maint and
password:password.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angus
ComberSent: Friday, March 25, 2005 8:08 AMTo:
asterisk-users@lists.digium.comSubject:
Type help-ahh from the console and you
will be able to change logins etc.
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angus Comber
Sent: Friday, March 25, 2005 10:08
AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] What
voip technocrat [EMAIL PROTECTED] wrote:
my aim is every call needs have wrapuptime of 5000 ms but when ever a
call comes its directly connecting not wating any more.
your views will be highly regarded
with regards
I'm using AddQueueMember.. for me, wrapuptime only seems to work from the
Hello,
I just setup the Asterisk to integrate with Panasonic legacy PBX. Config as followings,
PSTN -- PanasonicPBX--TDM400P(FXO)--AsteriskPC-- Internet
* is for AA / Voicemail and VOIP in/out
Currently the AA / Voicemail function for incoming PSTN callsare working well.
My problem is
On Fri, 25 Mar 2005 11:54:21 +0100, asterisk [EMAIL PROTECTED] wrote:
I have installed asterisk 1.05 on debian sarge (binary package)
with an I4l modem and 4 x-lite softphone and 2 SIP hardphones (Yuxin 100)
I am trying to get supervised/ attended tranfer working, blind transfer
by pressing
Type help-aah and you will
get list of commands to reset your passwords.
Neel
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angus Comber
Sent: Friday, March 25, 2005 7:08
AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] What is
web login
objective: users connected to box A can dial the extension number of
users connected to box B
boxA at location 1: works fine for internal lan users using the
firefly softphone
boxB at location 2: works fine for internal lan users using the
firefly softphone
Both the boxes have a IAX trunk
Login: maint
Password: password
-Kerry
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angus
ComberSent: Friday, March 25, 2005 7:08 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] What is web
login password for [EMAIL PROTECTED]
Hello
I have setup [EMAIL
We have some good walkthrus at http://www.geekgazette.com. These should
answer most of your questions all in one shot.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Trevor
Tregoweth
Sent: Friday, March 25, 2005 1:52 AM
To: 'Asterisk Users
I have searched both the wiki and googled looking for a solution to a square key
configuration. I need to have C.O. lines to appear on the buttons to facilitate
a small office. All of the users can see each other and calls are put on hold
and picked up by the other users instead of
Is it possible to rewrite caller id's?
I would like to have sip phones appear by their local cid
(like Henk 208) but when they call out using the PRI I would like their
full DID (MSN) to appear (like 0031201234567)
I could ofcourse set callerid to the main phonenumber but surely there
must be a
On Fri, 25 Mar 2005, Remco Barende wrote:
Is it possible to rewrite caller id's?
I would like to have sip phones appear by their local cid
(like Henk 208) but when they call out using the PRI I would like their
full DID (MSN) to appear (like 0031201234567)
I could ofcourse set callerid
Vikram Rangnekar wrote:
+++ Dan [20/03/05 09:17 +0200]:
Hi James,
- Original Message -
From: James Coberly [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, March 19, 2005 11:41 PM
Subject: Re: [Asterisk-Users]
I have the following config:
[app-callforward]
; dialed call forward app - forwards calling extension
exten = _*72.,1,DBput(CF/${CALLERIDNUM}=${EXTEN:3})
exten = _*72.,2,Answer
exten = _*72.,3,Wait(1)
exten = _*72.,4,Playback(loligo/call-fwd-unconditional)
exten = _*72.,5,Playback(loligo/for)
I would like to have sip phones appear by their local cid
(like Henk 208) but when they call out using the PRI I would like their
full DID (MSN) to appear (like 0031201234567)
I could ofcourse set callerid to the main phonenumber but surely there
must be a better solution?
Thanks!!
Remco
I set the
Greetings!
I have a FreeBSD 5.3 running on Intel SR1300 (dual xeon 2.6, scsi) server,
with ztdummy.ko driver as timing source for asterisk.
The typical output from zttest is:
$ zttest
Opened pseudo zap interface, measuring accuracy...
[..skip..]
--- Results after 192 passes ---
Best: 99.987793
http://www.identafone.com/cidpop.html Show right on the product page that it
uses asttapi and integrates with Asterisk
Henry
.
- Original Message -
From: Anton Krall [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent:
On Fri, 25 Mar 2005, Remco Barende wrote:
exten = _91NXXNXX,1,SetCallerID(IgLou Internet 5029663848)
exten = _91NXXNXX,2,Dial(Zap/r1/${EXTEN:1})
Yes, but this way you can only display one single phone number, and not
the MSN number for each SIP phone?
For example Henk has
On Friday 25 March 2005 16:30, Remco Barende wrote:
I would like to have sip phones appear by their local cid
(like Henk 208) but when they call out using the PRI I would like
their full DID (MSN) to appear (like 0031201234567)
I could ofcourse set callerid to the main phonenumber but
My home office is away from my house - so if anyone
rings door I cannot hear it. How would I rig up a doorbell which would
ring an extension on my Asterisk box?
Angus Comber
[EMAIL PROTECTED]
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Asterisk-Users mailing list
Jason Brown wrote:
| Anyone have experiece with polycom phones?
|
| I am experiencing a really weird problem. In an office where I have
| the following extensions:
| On the Polycom phones, when I want to dial from extension
100 to any
| extension 120 or above, or dial out, it dials just fine. If
I
Hi,
-Original Message-
I would like to have sip phones appear by their local cid
(like Henk 208) but when they call out using the PRI I
would like their
full DID (MSN) to appear (like 0031201234567)
I could ofcourse set callerid to the main phonenumber but
surely there
must
Hi,
-Original Message-
I would like to have sip phones appear by their local cid
(like Henk 208) but when they call out using the PRI I
would like their
full DID (MSN) to appear (like 0031201234567)
I could ofcourse set callerid to the main phonenumber but
surely there
must
Search Google. This is not a key system it is
a pbx. I don't think you can accomplish what you want with
this.
- Original Message -
From:
Mark W Wood
To: asterisk-users@lists.digium.com
Sent: Friday, March 25, 2005 10:00
AM
Subject: [Asterisk-Users] Square Key
This can be accomplished if the last 3 of the number you want to send to the
outside match the extension by using variables.
- Original Message -
From: Sean A. Newton [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
I would you an ATA and something like a Viking door
box. Then if they ring the door bell it can call your phone and you could
speak to the person to tell them you are on your way, leave the package or
whatever.
- Original Message -
From:
Angus
Comber
To:
I have an account with a IAX service provider that I'm happy with but
recently I started getting rather strange reports from users. They're
saying that occasionally, they'll be on a call via the provider and the
outbound audio appears to slowly fade out to nothing with a bit of static
during the
I have multiple locations running * where all the phone are
on their own lan and all the data is on a separate lan.
The problem is they are sharing the same dsl connection.
The locations are IAX2 trunked together, but it only takes
one data down/up load to just kill the voice.
What I am looking
hi all;
my first post/question is a bit vague. i'll be more specific on
[EMAIL PROTECTED] usage (eg: how was it?):
1.
am able to make it running at home using a braodband connection of
dynamic IP with no-ip, i able to SSH to the box and access to the web
pages. the problem is with the maint
hi all,
can any 1 pls tell me the context i shld add on sip.conf for
Audiocodec MP108 8 fxs please.
can`t get a dialtone only busy signal.
Thnx ppls
Imran
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
I have been playing with getting the sample.call file to work by dropping it into /var/spool/asterisk/outgoing. The process works to the point of calling the desired number and plays the message. The problem is that the message starts playing almost immediately, so if the called person takes 2 or
Remco Barende wrote:
For example Henk has SIP/208 and MSN 0031201208
I would like to display Henk 208 for any call that stays in the
company but 0031201208 to the outside.
If your internal numbers always match your outside numbers just prefix it
SetCallerID(Company Name
Here's our recent announcement of our new Asterisk Installation CD set:
Signate has announced its new Asterisk Installation 2005 CD Set. It's, a
complete software PBX (private branch exchange) telephony appliance in a single
package. The CD set installs Linux pre-configured for telephony, a
On Fri, 25 Mar 2005 04:07:13 -0500, Nabeel Jafferali
[EMAIL PROTECTED] wrote:
What happens if a SIP call is routed through more
than one * server?
If canreinvite=yes for all the peers involved, and t or T is not used in
the Dial command, then the audio would get routed directly between the
What about Callerid on call forwarding? I.e. an external call comes in
and is forwarded to a cell phone, how do I make the callerid that is
displayed on the cell phone the same as the inbound call?
Thanks,
Max
Max Clark
max [at] clarksys.com
http://www.clarksys.com
Remco Barende wrote:
Is
Linksys makes a VPN router with Dual WAN interfaces and QoS
http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-31672629504.htm
On Fri, 2005-03-25 at 11:13, Bob Knight wrote:
I have multiple locations running * where all the phone are
on their own lan and all the data is on a
[EMAIL PROTECTED] wrote on 03/25/2005 09:14:42 AM:
Hello
I want to to know if the motherboards VIA are fully supporte by
asterisk.
This is a complex question.
The *software* is fully supported. Depending on the CPU you use, you may
have to modify the makefiles (some VIA CPU's do not
[EMAIL PROTECTED] wrote on 03/25/2005 09:14:42 AM:
Hello
I want to to know if the motherboards VIA are fully supporte by
asterisk.
This is a complex question. The *software* runs on Mini-ITX (what I
assume you're asking about) just fine. The *hardware* *may* have issues
however.
These
On Fri, 25 Mar 2005, Bob Goddard wrote:
On Friday 25 March 2005 16:30, Remco Barende wrote:
I would like to have sip phones appear by their local cid
(like Henk 208) but when they call out using the PRI I would like
their full DID (MSN) to appear (like 0031201234567)
I could ofcourse set callerid
On 09:13, Fri 25 Mar 05, Bob Knight wrote:
I have multiple locations running * where all the phone are
on their own lan and all the data is on a separate lan.
The problem is they are sharing the same dsl connection.
The locations are IAX2 trunked together, but it only takes
one data down/up
http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20door
gee that took a lot of effort.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angus Comber
Sent: Friday, March 25, 2005 11:57
AM
To:
asterisk-users@lists.digium.com
Subject:
On Fri, 25 Mar 2005, Trevor Peirce wrote:
Remco Barende wrote:
For example Henk has SIP/208 and MSN 0031201208
I would like to display Henk 208 for any call that stays in the company
but 0031201208 to the outside.
If your internal numbers always match your outside numbers just prefix
On Fri, 25 Mar 2005 [EMAIL PROTECTED] wrote:
I have been playing with getting the sample.call file to work by dropping it
into
/var/spool/asterisk/outgoing. The process works to the point of calling the
desired
number and plays the message. The problem is that the message starts playing
Jermal,
Your second round of questions are just as basic, do some research on
wiki.
1/ did you not see that when you log onto the console it says type
help-aah to change passwords?
2/ [EMAIL PROTECTED] doesn't need any more documentation - all of the
documentation for [EMAIL PROTECTED] is on the
Andrew Kohlsmith wrote:
Also, carrier access has an incredible support site that they do not charge
for. You do need to register with them but that's free.
Just for a follow up to this statement,
I recently purchased an Adit 600 via eBay, tried to gain access to
Carrier Access's website.
Just found a 12 port single card with opensource drivers
12 user configurable FX0/FXS analogue ports for $1,680 at asterisk mall
( http://www.asteriskmall.com ).
I am not sure how well this card works with asterisk. Has anyone used
these cards?
Voip supply has a few 24 port gateways that are
SNIP
SigMON, Signate's included PBX monitoring software,
helps keep the PBX running.
SigMON monitors about 20 different conditions on the PBX
and sends alerts if a
condition needs to be attended to. Monitored conditions
range from hardware
conditions such as available disk space and CPU
Sorry for my bad english. I'm a french guy.
Absolument rien à critiquer de ton anglais
I have the same problems with siemens dect phones S100
The caller id don't work on tdm...
snip
Try adding
cadence=250,1500,1500,3000,1500,3000
In zapata.conf
And use in extension.conf
exten =
CAUTION: voicemail screwed up for me (garbled) with upgrade to 23, went
back to .22 and all is well.
Don't know why, I'll look at it later.
dean collins wrote:
Version 1.0.5.23 is now available from http://gs-firmware.gratissip.dk/
Or directly from Grandstream at
Hi,The lan is probably not the problem, but the dsl connection is.There are some things you can do that can help to a certain degree.First, set tos=lowdelay in your iax.conf.Most routers obey the ToS field.second, try to find out if your firewall (I assume you use one) support QoS.then there is a
Date: Fri, 25 Mar 2005 18:39:26 +0100 (CET)
From: Peter Svensson [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Zap Detect called party pickup
On Fri, 25 Mar 2005 [EMAIL PROTECTED] wrote:
I have been playing with getting the sample.call file to work by
dropping it into
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