9V DC, 1500mA Regulated
Tip positive, 5.0mm outer-diameter
2.5mm inner-diameter connected
http://www.digium.com/downloads/product_sheets/IAXy.pdf
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Friday, August 27, 2004 4:54 PM
To:
You know, if you purchased the kit from Digium it includes support
direct from them. Especially if you're desparate for help.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of steve
Sent: Wednesday, August 25, 2004 12:03 AM
To: [EMAIL PROTECTED]
Subject:
It might make sense for * to parse the register line from right to left.
Then it wouldn't be an issue. Or am I missing another issue that would
arise?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg Hill
Sent: Monday, August 16, 2004 1:21 PM
To:
Why hack the code for this? Just implement a wait() in your dialplan.
That way you can switch back and forth between outbound-only and in/out
by just changing the wait(120) to wait(1).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Cook
Sent:
I guess the question to ask is... Is the macro function designed to
execute one extension logic and then exit back to it's original context,
or is it designed to allow you to run multiple extension logics before
kicking back?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Check your config file. the 't' doesn't stand for terminate. It stands
for timeout
http://www.voip-info.org/wiki-Asterisk+t+extension
Try adding your operator to the 't' extension instead of hanging up on
them.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
It doesn't look like you have a context set for phone1. Try putting
context=sip in the phone1 section like you have in phone2. That'll put
both in the same context of your extensions.conf file and should allow
interaction between the two.
-Original Message-
From: [EMAIL PROTECTED]
Just use a standard phone cable. It will seat properly.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Florin
Andrei
Sent: Friday, July 16, 2004 5:12 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] PSTN/phone/FXO/FXS cabling issue
I just received a
You might want to try removing the hyphen. It could be misinterpreting
it? Might want to try simplifying things a bit too for testing
purposes. Take out the PSTN-1 and put in the ZAP/1 directly into your
dial plan to verify that * can access the ZAP channel correctly.
-Original
Using the EXTEN variable will give you the extension that was dialed. Try using
CALLERIDNUM (for this problem and your other post).
http://www.voip-info.org/wiki-Asterisk+variables
From: [EMAIL PROTECTED] on behalf of Paul Mahler
Sent: Fri 5/14/2004 1:47 PM
It's obvious you've at least tried to figure it out since you've used
the SetMusicOnHold app, so I'll be nice. Try MusicOnHold()
http://www.voip-info.org/wiki-Asterisk+cmd+MusicOnHold
-Original Message-
From: leonardo [mailto:[EMAIL PROTECTED]
Sent: Sunday, May 09, 2004 9:03 AM
To:
Welcome to the wonderful world of Asterisk! In the future, you might
want to make sure that you post in plain text mode instead of HTML.
There are quite a few people here who are great assets that won't even
read if you post in HTML.
Your problem has to do with the contexts. In your zapata.conf
However, if there is no answer, or the extension is busy, *
just keeps on trying to connect, and never drops to voicemail
(busy or unavailable).
exten = _7XX,1,Dial(zap/1/${EXTEN}|5m)
try something like exten = _7XX,1,Dial(zap/1/${EXTEN},20) where 20 is
the number of seconds you want it to
http://www.voip-info.org/wiki-Asterisk+codecs
G.723.1 can only be used in pass-thru mode.
-Original Message-
From: Todd Wallace [mailto:[EMAIL PROTECTED]
Sent: Monday, April 12, 2004 12:48 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] G.723
Is it at all possible?
If I remember correctly (and I could be wrong) I think you have to
answer the line first...
exten = s,1,Answer
exten = s,2,Zapateller(nocallerid)
exten = s,3,Privacymanager
exten = s,4,Dial(a bunch of SIP extensions)
-Original Message-
From: Mark Phillips [mailto:[EMAIL PROTECTED]
Apr 11 08:59:27 NOTICE[81926]: chan_sip.c:5568 handle_request: Registration from
'sip:[EMAIL PROTECTED]' failed for '192.168.0.6'
Are you sure your phone isn't registering? These errors aren't related to your
grandstream. Do a sip show peers at the Asterisk CLI and see if it shows your phone
Sounds like an error in your config file. Want to paste the contents
in? Thanks...
Sean
-Original Message-
From: Jeremy Bogan [mailto:[EMAIL PROTECTED]
Sent: Wednesday, April 07, 2004 8:06 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] MySQL CDR
Hi,
I'm trying to get CDR
What happens when you do stop now like the error states?
Sean
-Original Message-
From: Ryan Parlee [mailto:[EMAIL PROTECTED]
Sent: Saturday, April 03, 2004 9:56 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Unabled to exit console
No matter what I try, Asterisk won't let me out
just do -vvvr
-Original Message-
From: Ryan Parlee [mailto:[EMAIL PROTECTED]
Sent: Saturday, April 03, 2004 11:39 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Unabled to exit console
Okay, but if I do
/usr/sbin/asterisk
Then when I connect, using -r I don't get
Title: Message
JR,
This
is the third time you've posted this same information. We are all glad
that you're contributing to the community, but not over and over! Also,
you might want to add this to the Wiki if you already haven't.
Thanks,
Sean
-Original Message-From: JR
it is not included with the asterisk distribution. you must download it
separately. asterisk_addons.
-Original Message-
From: Jorge de J. Ramirez S. [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 30, 2004 2:10 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] mysql or postgresql?
The answer is in the error use FXS signalling. replace fxo_ks with
fxs_ks.
Sean
-Original Message-
From: vozip [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 30, 2004 2:31 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] problem with configuration.
Importance: High
Hi,
have you installed the mysql-devel package?
-Original Message-
From: Jorge de J. Ramirez S. [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 30, 2004 6:11 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] RE: mysql or postgresql?
thanks for awnser, I've already download from CVS the
Title: Message
You
could use the t extension to accomplish this. But if you're happy with
your way... :-)
Sean
-Original Message-From: Gene Kochanowsky
[mailto:[EMAIL PROTECTED] Sent: Tuesday, March 30, 2004
8:53 PMTo: [EMAIL PROTECTED]Subject: RE:
[Asterisk-Users]
Try doing an answer first:
exten = 8600,1,Answer
exten = 8600,2,Meetme,1234
Might also be worth doing a Meetme(1234) instead of Meetme,1234. I
believe both should work, but..
-Original Message-
From: Mailling LIst [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 25, 2004 3:17 PM
A quick search of Yahoo found quite a few reports of issues in various
devices with spaces in the SSID. Seems a lot of implementations fail to
properly handle the space. Definitely sounds like a WiSIP issue, but
might be worth removing the space from your SSID if at all
convenient
Sean
Make sure that the context specified in the zapata.conf section for
ZAP/1 actually exists in your extensions.conf and isn't blank. I had
the same problem with two X100P's in my system a few weeks ago. Hope
this helps...
Sean
-Original Message-
From: Chris Clifton [mailto:[EMAIL
1. There are VERY clear directions at the bottom of every email on how
to unsunscribe.
2. If you MUST send this to the list, make sure you spell unsubscribe
right.
-Original Message-
From: M Q [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 23, 2004 10:02 PM
To: [EMAIL PROTECTED]
Come on, man! Take a look at all of the wonderful resources available
before asking questions. http://www.voip-info.org is your friend.
Start there, and take a few days to read over everything. Then you will
find this: http://www.voip-info.org/wiki-Asterisk+Hardware. The mailing
list is a
Call Digium. They're ~$60 each.
Sean
-Original Message-
From: Wilson Pickett [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 10, 2004 10:25 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] TDM400P - upgradable how?
Hi,
I ordered the * Developers Kit which I though would be able
If you don't like smart-assed replies, I'd recommend you try to answer
your own questions first. It is a very easy thing to test. As a matter
of fact, the demo config files do exactly this. No one here is out to
slam you personally. Don't take it that way. As a matter of fact,
everyone is
Hi all Sorry for the last post! Not enough sleep combined with inattention
caused me to reply to the wrong message.
Sean
-Original Message-
From: Anton Tinchev [mailto:[EMAIL PROTECTED]
Sent: Mon 2/23/2004 12:25 AM
To: [EMAIL PROTECTED]
There are issues with SIP channels. It automatically creates SIP
objects with the unique four digit identifier on the end. I remember
reading about this, but don't remember any solution besides get used
to it!
Sean
-Original Message-
From: Lenny Tropiano / asterisk.org Mailing list
It shows that way on my RH9 box, but that is not what's causing your
problems with the drivers not seeing the card. Have you configured your
zaptel.conf for your hardware? Have you done ztcfg -vv? What order did
you modprobe? We need a little more info to help Thanks!
Sean
-Original
Now we're getting somewhere! The TDM400P is a PCI 2.2 card. So
depending on what you mean by an older motherboard, that might be your
problem.
-Original Message-
From: Tim Sailer [mailto:[EMAIL PROTECTED]
Sent: Friday, February 06, 2004 6:53 PM
To: [EMAIL PROTECTED]
Subject: Re:
I believe it is a requirement. When I bought mine, I had the same
issue. After talking to Digium, I was informed that the card would not
be recognized in a non-PCI 2.2 slot. I put it in another (newer) box
and it came right up.
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL
Notice he did indicate he installed from the rpm's, so he's not using
the source. But I agree on the lazy part! There are tons of resources
available. Try http://www.voip-info.org and look at the config file
section. Then try to create what you need (they're not hard for
proof-of-concept
do you have the kernel source installed?
-Original Message-
From: Tim Sailer [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 05, 2004 10:33 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] zaptel on Debian
Does anyone have the zaptel modules built for Debian 2.4.24 kernel?
When I
sounds like you need to do some reading at the many fine resources
available. start at http://www.voip-info.org. Here's a hint for you
though
exten = s,1,Answer
exten = s,2,VoicemailMain
Barring that, just run 'make samples' which will create a wonderful set
of sample config files which
well quit with the suspense already and tell us who! :-)
-Original Message-
From: Rob Fugina [mailto:[EMAIL PROTECTED]
Sent: Saturday, January 31, 2004 7:56 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] 8 lines - best approach
On Sat, Jan 31, 2004 at 12:00:49PM -0800,
I'm not sure if you're trying to accomplish something specifically by using rc.local,
but I use RH9, and I used make config on both asterisk and zaptel and that created the
correct init files for me. Starts up perfect every time!
Sean
-Original Message-
From: listas iPfone
funny... I got an immediate response, and within 1 hour had my account
activated. and this was today.
-Original Message-
From: Chris Albertson [mailto:[EMAIL PROTECTED]
Sent: Sunday, January 25, 2004 10:36 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Has Nufone gone
Title: Message
they
are 3rd-party. I bought one, and I bought one directly from Digium.
They both work the same as near as I can tell!
-Original Message-From: SamW
[mailto:[EMAIL PROTECTED] Sent: Wednesday, January 21, 2004 4:34
PMTo: [EMAIL PROTECTED]Subject:
for the record, mine has the same fcc id number as the Digiums. Is this
typical for copied hardware, or is there something a little fishy going
on here?
-Original Message-
From: Doug Meredith [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 21, 2004 7:20 PM
To: [EMAIL PROTECTED]
How do we know they're pirated? And how is a 132% difference in price
trivial? ($43 vs. $99.95). Don't get me wrong I have bought
hardware from Digium, and am very happy with all of it. But I also
purchased some SIP equipment and channel banks, all out of my own money.
I wanted to play
-
From: Dustin Goodwin [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 21, 2004 11:57 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Re: Digium X100P for $43
It's funny this is the second hardware counterfeiting story I have heard
this week. What is going on?
- Dustin -
Sean
You need a license for each end.
-Original Message-
From: Dinesh [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 20, 2004 8:41 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] RE: G729 question
Hi All,
I am looking into setting up my asterisk server in the next couple of
days,
Title: Message
sounds
like you're doing an 'asterisk -r' when it's not already running. try
'asterisk -vc' and see if it launches. The more v's, the more verbose the
output.
-Original Message-From: listas iPfone
[mailto:[EMAIL PROTECTED] Sent: Friday, January 16, 2004 3:48
I am having problems too Just shy of the 5-second mark in the
test vm.
WMP 9.00.00.3075
Windows 2000 SP4
-Original Message-
From: Warwick Ward-Cox [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 15, 2004 10:57 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] wav49
Actually he found it in the dumpster after the police threw it out
following a bust! Does anyone want to send a dollar to Mr. Happy?!
-Original Message-
From: C. Maj [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 15, 2004 12:40 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] *
well, it does say SIMPLEX in the fxp0 flags section. I don't honestly
know if this means it's negotiated half duplex, or something beyond
that 10baseT is capable of running full duplex, although this
requires a NIC capable of is, as well as a switch that can do FD. And
regarding the 1%
knot n. A unit of speed, one nautical mile per hour
thanks to our good friends at reference.com. Are we done yet?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Monday, January 12, 2004 10:10 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] More words
have you tried:
access-list 61 permit 10.1.1.2 0.0.0.0
I'm not 100% sure that the mask is implied if you don't specify it. And
with Cisco ACL's, the mask is the inverse of the standard IP mask.
-Original Message-
From: B. J. Bomar [mailto:[EMAIL PROTECTED]
Sent: Monday, January 12,
just drop it! it is for them to iron out! and for the record, I received my order
within a week of placing the order.
-Original Message-
From: admin [mailto:[EMAIL PROTECTED]
Sent: Sat 1/10/2004 3:23 PM
To: [EMAIL PROTECTED]
Cc:
time to take this off-list.
-Original Message-
From: mattf [mailto:[EMAIL PROTECTED]
Sent: Saturday, January 10, 2004 10:05 PM
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Chagres Technologies, Inc
Hello,
I have the shipping numbers for the first 2 shipments of 40 phones
Am I missing something? Is there another way to pipe large quantities
of analog lines (FXS or FXO) into *? Seriously, is there another way?
Sean
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: Saturday, January 10, 2004 9:48 PM
To: [EMAIL PROTECTED]
Subject: Re:
There is...
#include filename
-Original Message-
From: Lion Templin [mailto:[EMAIL PROTECTED]
Sent: Fri 1/9/2004 7:14 PM
To: [EMAIL PROTECTED]
Cc:
Subject: [Asterisk-Users] file_inlcude .. why not?
Don't
you can always do a restart when convenient within asterisk, and it
will do it's thing when all lines are clear
-Original Message-
From: Jonathan Moore [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 08, 2004 12:31 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Re: 911 and
From a hardware standpoint, each PCI slot is numbered. The chipset of each system
board determines what the order is. For example, you have a T100P in what the system
knows as PCI4 and no other T100P's in the system the software will see this as the
first T100P device. But if you add another
I don't think he asked for anyone to validate his question! Man! Just because it
might not be valid for you doesn't mean it isn't a valid question!
Sean
-Original Message-
From: Michael Welter [mailto:[EMAIL PROTECTED]
Sent: Wed 1/7/2004 6:51 PM
To: [EMAIL PROTECTED]
Cc:
Then all you need is a cheap way to integrate SIP, h323, and all the other advanced
features that * brings to the table
-Original Message-
From: James H. Thompson [mailto:[EMAIL PROTECTED]
Sent: Wed 1/7/2004 9:30 PM
To: [EMAIL PROTECTED]
have you set up the db schema? and have you entered any sip data into the db?
Sean
-Original Message-
From: Chandra [mailto:[EMAIL PROTECTED]
Sent: Tue 1/6/2004 10:57 PM
To: [EMAIL PROTECTED]
Cc:
Subject: [Asterisk-Users] no results.
and data... but i have something
like 4 datas in my sip table
1234,account,sip1,0
1235,account,sip2,0
1236,user,sip3,0
1236,peer,sip3,0
what do u mean by db schema???
- Original Message -
From: Sean
just make that very
unclear!
Sean
-Original Message-From: Sean Cheesman
[mailto:[EMAIL PROTECTED] On Behalf Of Sean
CheesmanSent: Tuesday, January 06, 2004 11:43 PMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users] no
results.
the database schema is the table and it's
my biggest concern about defaulting the context to anything at all
besides [default] is that you then have to remember to configure the
voicemail.conf with the corresponding contexts. as it stands, you have
the ability to do just that, but you don't have to. if you have several
hundred
Hi Jess,
It looks like your problem is with the extension increment. If there is no answer in
the allotted time, the count increses by one. If the line is busy, the count
increases by 101. Also, have you actually created the vm boxes you're referencing?
Thanks!
Sean
both of your messages have shown up John. It's just running a little slow today
Sean
-Original Message-
From: John Coll [mailto:[EMAIL PROTECTED]
Sent: Mon 1/5/2004 5:23 PM
To: [EMAIL PROTECTED]
Cc:
Subject: [Asterisk-Users] Are
There are no guarantees that the voicemail will be in the same context
as the extension. By giving you the ability and flexibility of defining
everything independently, there's not much you can't do! Remember, the
context call in the sip.conf refers to the context in extensions.conf.
the
Hi John,
Try adding username=5702 and username=5703 to each of the configs in
sip.conf. I recall I had this problem with the Grandstreams.
-Original Message-
From: John Coll [mailto:[EMAIL PROTECTED]
Sent: Saturday, January 03, 2004 11:56 AM
To: [EMAIL PROTECTED]
Subject: RE:
There are many people on this list that are more than happy to help you with
a problem if you know how to ask the question. But if you've tried to keep
up with this mailing list over any amount of time, you will see how quickly
it becomes frustrating when people ask the same questions over and
shipping.conf
Of course, this is only one of many ways you could use the #include
function!
Sean
-Original Message-
From: Lance Arbuckle [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 30, 2003 4:54 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] include a file ?
Sean
Hi Lance,
Watch your voicemail-busy line. The step count looks like it's wrong. It's
never fun to track down a little problem like that!
Sean
-Original Message-
From: Lance Arbuckle [mailto:[EMAIL PROTECTED]
Sent: Monday, December 29, 2003 6:48 PM
To: [EMAIL PROTECTED]
Subject: Re:
A
search on Yahoo brought up quite a few RJ45-BNC cable
sets
-Original Message-From: Hector Q.-datafull
[mailto:[EMAIL PROTECTED]Sent: Sunday, December 28, 2003 5:20
PMTo: [EMAIL PROTECTED]Subject:
[Asterisk-Users] Digium Wildcat E100 card mechanics issue
Hello,
I think what Steve was getting at was interrupt sharing. Is the fxs card on
the same interrupt as anything else?
Sean
-Original Message-
From: Victor Rini [mailto:[EMAIL PROTECTED]
Sent: Sunday, December 28, 2003 10:21 PM
To: '[EMAIL PROTECTED]'
Subject: [Asterisk-Users] RE: TDM Card
now we're getting somewhere! anything above interrupt 15 will be interrupt
sharing. bad! If you can get the cards assigned to 10 or 11, you should be
in better shape.
Sean
-Original Message-
From: Victor Rini [mailto:[EMAIL PROTECTED]
Sent: Monday, December 29, 2003 12:12 AM
To:
You can check it out via CVS. asterisk-addons
-Original Message-
From: David A. Lauer [mailto:[EMAIL PROTECTED]
Sent: Saturday, December 27, 2003 3:16 PM
To: Asterisk Users
Subject: [Asterisk-Users] mysql cdrs
How can I download the asterisk-addons and setup CDR support for mysql?
I
in the simplest terms, ztcfg takes your zaptel.conf, parses it, and lets
asterisk know what hardware you have and how it's configured.
-Original Message-
From: Ing. Angel Gomez Garcia [mailto:[EMAIL PROTECTED]
Sent: Friday, December 26, 2003 7:58 PM
To: Asterisk Users
Subject:
voicemail notification?
-Original Message-From: bam
[mailto:[EMAIL PROTECTED]Sent: Wednesday, December 24, 2003 12:17
PMTo: [EMAIL PROTECTED]Subject:
[Asterisk-Users] Grandstream 102 flashing displayThe
phone powers up and I can make calls through my Asterisk gateway to
I'm going to take a stab at this, so someone correct me if I'm wrong! If
you're calling one g729 device from another, the call is actually handed off
without any decoding done, therefore the licensing isn't needed. If * has
to connect the g729 call to another format, then the licensing comes in
The problem occurs when the software is expecting the packet in a certain
timeframe so that it can reassemble it in a timely manner. It's not a big
deal with a web page or something along that lines. But when a voice
application cannot get reassembled in a timely manner, you'll surely notice
it!
Telecommunications Device for the Deaf
-Original Message-
From: Philipp von Klitzing
[mailto:[EMAIL PROTECTED]
Sent: Monday, December 22, 2003 11:13 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] ToIP (TDD over IP)
Hi!
I'm also curious if anyone else is doing this or if anyone
You have SIP/lcs-sipura1 listed for both extensions in your extensions.conf.
Is this a type-o in your email?
-Original Message-
From: Ariel Batista [mailto:[EMAIL PROTECTED]
Sent: Monday, December 22, 2003 1:11 PM
To: Asterisk User List
Subject: [Asterisk-Users] Sipura 2000 configuration.
I might be wrong, but isn't is just saying that the packet has been delayed
x-ms? I'm not sure it's saying that Packet 52 arrived 5ms after packet 51.
Although even if it was, that doesn't mean that it was sent 5ms after packet
51 either.
-Original Message-
From: Andrew Kohlsmith
Hi all,
I am looking at setting up a TDMoE link between * boxes and am having a
rough time locating and documentation or configuration examples. I have
gotten far enough to get the dynamic link up between boxes, but not sure
where to go from here. I'm not even sure which modules need to be
After searching the archives for a while, I couldn't find any easy way to
get everything loaded on startup. So I decided to take a stab at writing
some notes on what I've found. If everyone chips in, maybe we can make that
part easier for new users!
Both the Zaptel and Asterisk packages have a
Might be a stupid question, but is there a default gateway set on the 7960?
-Original Message-
From: Paul Mahler [mailto:[EMAIL PROTECTED]
Sent: Thursday, December 18, 2003 7:04 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisco 7960 - can't traverse NAT?
I have a 7960 running
: Re: [Asterisk-Users] Trunk Groups and Multiple Asterisk
Machines
At 7:44 PM -0500 12/17/03, Sean Cheesman wrote:
Hello all,
I have no problems setting up trunk groups in general, but is there a way
to
set up a trunk group for outbound calls that includes channels on multiple
servers? I might have
try adding username=1005 under [1005] and see if that helps
-Original Message-
From: PBX [mailto:[EMAIL PROTECTED]
Sent: Thursday, December 18, 2003 10:42 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SIP / X-ten Softphone
I know this has been covered more times than to mention
Hello all,
I have no problems setting up trunk groups in general, but is there a way to
set up a trunk group for outbound calls that includes channels on multiple
servers? I might have missed something somewhere, but I couldn't find any
reading about this topic. Thanks!
Sean
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